1 ==============================================================================
3 === This file documents the new and/or enhanced functionality added in
4 === the Asterisk versions listed below. This file does NOT include
5 === changes in behavior that would not be backwards compatible with
6 === previous versions; for that information see the UPGRADE.txt file
7 === and the other UPGRADE files for older releases.
9 ==============================================================================
10 ------------------------------------------------------------------------------
11 --- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
12 ------------------------------------------------------------------------------
19 * The application no longer does agent authentication. The dialplan needs to
20 perform this function before running AgentLogin. If the agent is already
21 logged in, dialplan will continue with the AGENT_STATUS channel variable
22 set to ALREADY_LOGGED_IN.
26 * Application removed. It was a holdover from when AgentCallbackLogin was
31 * All participants in a bridge can now be kicked out of a conference room
32 by specifying the channel parameter as 'all' in the ConfBridge kick CLI
33 command, i.e., "confbridge kick <conference> all"
37 * ForkCDR no longer automatically resets the forked CDR. See the 'r' option
40 * Variables are no longer purged from the original CDR. See the 'v' option for
43 * The 'A' option has been removed. The Answer time on a CDR is never updated
46 * The 'd' option has been removed. The disposition on a CDR is a function of
47 the state of the channel and cannot be altered.
49 * The 'D' option has been removed. Who the Party B is on a CDR is a function
50 of the state of the respective channels, and cannot be altered.
52 * The 'r' option has been changed. Previously, ForkCDR always reset the CDR
53 such that the start time and, if applicable, the answer time was updated.
54 Now, by default, ForkCDR simply forks the CDR, maintaining any times. The
55 'r' option now triggers the Reset, setting the start time (and answer time
56 if applicable) to the current time.
58 * The 's' option has been removed. A variable can be set on the original CDR
59 if desired using the CDR function, and removed from a forked CDR using the
62 * The 'T' option has been removed. The concept of DONT_TOUCH and LOCKED no
63 longer applies in the CDR engine.
65 * The 'v' option now prevents the copy of the variables from the original CDR
66 to the forked CDR. Previously the variables were always copied but were
67 removed from the original. Removing variables from a CDR can have unintended
68 side effects - this option allows the user to prevent propagation of
69 variables from the original to the forked without modifying the original.
73 * Added the 'n' option to MeetMe to prevent application of the DENOISE function
74 to a channel joining a conference. Some channel drivers that vary the number
75 of audio samples in a voice frame will experience significant quality problems
76 if a denoiser is attached to the channel; this option gives them the ability
77 to remove the denoiser without having to unload func_speex.
81 * The NoCDR application is deprecated. Please use the CDR_PROP function to
83 * While the NoCDR application will prevent CDRs for a channel from being
84 propagated to registered CDR backends, it will not prevent that data from
85 being collected. Hence, a subsequent call to ResetCDR or the CDR_PROP
86 function that enables CDRs on a channel will restore those records that have
87 not yet been finalized.
91 * Add queue available hint. exten => 8501,hint,Queue:markq_avail
92 Note: the suffix '_avail' after the queuename.
93 Reports 'InUse' for no logged in agents or no free agents.
94 Reports 'Idle' when an agent is free.
96 * The configuration options eventwhencalled and eventmemberstatus have been
97 removed. As a result, the AMI events QueueMemberStatus, AgentCalled,
98 AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be
99 sent. The "Variable" fields will also no longer exist on the Agent* events.
101 * The queue log now differentiates between blind and attended transfers. A
102 blind transfer will result in a BLINDTRANSFER message with the destination
103 context and extension. An attended transfer will result in an
104 ATTENDEDTRANSFER message. This message will indicate the method by which
105 the attended transfer was completed: "BRIDGE" for a bridge merge, "APP"
106 for running an application on a bridge or channel, or "LINK" for linking
107 two bridges together with local channels.
109 * Queues now support a hint for member paused state. The hint uses the form
110 'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name}
111 are the name of the queue and the name of the member to subscribe to,
112 respectively. For example: exten => 8501,hint,Queue:sales_pause_mark.
113 Members will show as In Use when paused.
117 * The 'e' option has been deprecated. Use the CDR_PROP function to re-enable
118 CDRs when they were previously disabled on a channel.
119 * The 'w' and 'a' options have been removed. Dispatching CDRs to registered
120 backends occurs on an as-needed basis in order to preserve linkedid
121 propagation and other needed behavior.
125 * This application is deprecated in favor of the CHANNEL function.
129 * UserEvent will now handle duplicate keys by overwriting the previous value
130 assigned to the key. UserEvent invocations will also be distributed to any
131 interested res_stasis applications.
136 * Asterisk now optionally uses libxslt to improve XML documentation generation
137 and maintainability. If libxslt is not available on the system, some XML
138 documentation will be incomplete.
143 * Redirecting reasons can now be set to arbitrary strings. This means
144 that the REDIRECTING dialplan function can be used to set the redirecting
145 reason to any string. It also allows for custom strings to be read as the
146 redirecting reason from SIP Diversion headers.
148 * For DTMF blind and attended transfers, the channel variable TRANSFER_CONTEXT
149 must be on the channel initiating the transfer to have any effect.
151 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel
152 driver specific. If the channel variable is set on the transferrer channel,
153 the sound will be played to the target of an attended transfer.
155 * The channel variable BRIDGEPEER becomes a comma separated list of peers in
156 a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers
157 listed. Any more peers in the bridge will not be included in the list.
158 BRIDGEPEER is not valid in holding bridges like parking since those channels
159 do not talk to each other even though they are in a bridge.
161 * The channel variable BRIDGEPVTCALLID is only valid for two party bridges
162 and will contain a value if the BRIDGEPEER's channel driver supports it.
164 * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is
165 removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that
166 activated the dynamic feature.
168 * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set
169 only on the channel executing the dynamic feature. Executing a dynamic
170 feature on the bridge peer in a multi-party bridge will execute it on all
171 peers of the activating channel.
173 * A channel variable ATTENDEDTRANSFER is now set which indicates which channel
174 was responsible for an attended transfer in a similar fashion to
177 AMI (Asterisk Manager Interface)
179 * The SIPshowpeer action will now include a 'SubscribeContext' field for a peer
180 in its response if the peer has a subscribe context set.
182 * The SIPqualifypeer action now acknowledges the request once it has established
183 that the request is against a known peer. It also issues a new event,
184 'SIPQualifyPeerDone', once the qualify action has been completed.
186 * The PlayDTMF action now supports an optional 'Duration' parameter. This
187 specifies the duration of the digit to be played, in milliseconds.
189 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
190 updates when changes occur instead of requiring the use of pollmailboxes.
192 * CLI Command 'Manager Show Commands' no longer truncates command names longer
193 than 15 characters and no longer shows authorization requirement for commands.
194 'Manager Show Command' now displays the privileges needed for using a given
195 manager command instead.
197 * Added new action "ControlPlayback". The ControlPlayback action allows an AMI
198 client to manipulate audio currently being played back on a channel. The
199 supported operations depend on the application being used to send audio to
200 the channel. When the audio playback was initiated using the ControlPlayback
201 application or CONTROL STREAM FILE AGI command, the audio can be paused,
202 stopped, restarted, reversed, or skipped forward. When initiated by other
203 mechanisms (such as the Playback application), the audio can be stopped,
204 reversed, or skipped forward.
206 * Channel related events now contain a snapshot of channel state, adding new
207 fields to many of these events.
209 * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
210 in a future release. Please use the common 'Exten' field instead.
212 * The AMI event 'UserEvent' from app_userevent now contains the channel state
213 fields. The channel state fields will come before the body fields.
215 * The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
216 'UnParkedCall' have changed significantly in the new res_parking module.
218 The 'Channel' and 'From' headers are gone. For the channel that was parked
219 or is coming out of parking, a 'Parkee' channel snapshot is issued and it
220 has a number of fields associated with it. The old 'Channel' header relayed
221 the same data as the new 'ParkeeChannel' header.
223 The 'From' field was ambiguous and changed meaning depending on the event.
224 for most of these, it was the name of the channel that parked the call
225 (the 'Parker'). There is no longer a header that provides this channel name,
226 however the 'ParkerDialString' will contain a dialstring to redial the
227 device that parked the call.
229 On UnParkedCall events, the 'From' header would instead represent the
230 channel responsible for retrieving the parkee. It receives a channel
231 snapshot labeled 'Retriever'. The 'from' field is is replaced with
234 Lastly, the 'Exten' field has been replaced with 'ParkingSpace'.
236 * The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar
237 fashion has changed the field names 'StartExten' and 'StopExten' to
238 'StartSpace' and 'StopSpace' respectively.
240 * The deprecated use of | (pipe) as a separator in the channelvars setting in
241 manager.conf has been removed.
243 * Channel Variables conveyed with a channel no longer contain the name of the
244 channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now
245 ChanVariable: bar=baz. When multiple channels are present in a single AMI
246 event, the various ChanVariable fields will contain a suffix that specifies
247 which channel they correspond to.
249 * The NewPeerAccount AMI event is no longer raised. The NewAccountCode AMI
250 event always conveys the AMI event for a particular channel.
252 * All "Reload" events have been consolidated into a single event type. This
253 event will always contain a Module field specifying the name of the module
254 and a Status field denoting the result of the reload. All modules now issue
255 this event when being reloaded.
257 * The "ModuleLoadReport" event has been removed. Most AMI connections would
258 fail to receive this event due to being connected after modules have loaded.
259 AMI connections that want to know when Asterisk is ready should listen for
260 the "FullyBooted" event.
262 * app_fax now sends the same send fax/receive fax events as res_fax. The
263 "FaxSent" event is now the "SendFAX" event, and the "FaxReceived" event is
264 now the "ReceiveFAX" event.
266 * The MusicOnHold event is now two events: MusicOnHoldStart and
267 MusicOnHoldStop. The sub type field has been removed.
269 * The JabberEvent event has been removed. It is not AMI's purpose to be a
270 carrier for another protocol.
272 * The Bridge Manager action's Playtone header now accepts more fine-grained
273 options. "Channel1" and "Channel2" may be specified in order to play a tone
274 to the specific channel. "Both" may be specified to play a tone to both
275 channels. The old "yes" option is still accepted as a way of playing the
276 tone to Channel2 only.
278 * The AMI 'Status' response event to the AMI Status action replaces the
279 BridgedChannel and BridgedUniqueid headers with the BridgeID header to
280 indicate what bridge the channel is currently in.
282 * The AMI 'Hold' event has been moved out of individual channel drivers, into
283 core, and is now two events: Hold and Unhold. The status field has been
286 * The AMI events in app_queue have been made more consistent with each other.
287 Events that reference channels (QueueCaller* and Agent*) will show
288 information about each channel. The (infamous) "Join" and "Leave" AMI
289 events have been changed to "QueueCallerJoin" and "QueueCallerLeave".
291 * The MCID AMI event now publishes a channel snapshot when available and
292 its non-channel-snapshot parameters now use either the "MCallerID" or
293 "MConnectedID" prefixes with Subaddr*, Name*, and Num* suffixes instead
294 of "CallerID" and "ConnectedID" to avoid confusion with similarly named
295 parameters in the channel snapshot.
297 * The AMI events "Agentlogin" and "Agentlogoff" have been renamed
298 "AgentLogin" and "AgentLogoff" respectively.
300 * The "Channel" key used in the "AlarmClear", "Alarm", and "DNDState" has been
301 renamed "DAHDIChannel" since it does not convey an Asterisk channel name.
303 * "ChannelUpdate" events have been removed.
305 * AMI events now contain a SystemName field, if available.
307 * Local channel optimization is now conveyed in two events:
308 LocalOptimizationBegin and LocalOptimizationEnd. The Begin event is sent
309 when the Local channel driver begins attempting to optimize itself out of
310 the media path; the End event is sent after the channel halves have
311 successfully optimized themselves out of the media path.
313 * Local channel information in events is now prefixed with "LocalOne" and
314 "LocalTwo". This replaces the suffix of "1" and "2" for the two halves of
315 the Local channel. This affects the LocalBridge, LocalOptimizationBegin,
316 and LocalOptimizationEnd events.
318 * The option 'allowmultiplelogin' can now be set or overriden in a particular
319 account. When set in the general context, it will act as the default
320 setting for defined accounts.
322 * The 'BridgeAction' event was removed. It technically added no value, as the
323 Bridge Action already receives confirmation of the bridge through a
324 successful completion Event.
326 * The 'BridgeExec' events were removed. These events duplicated the events that
327 occur in the Briding API, and are conveyed now through BridgeCreate,
328 BridgeEnter, and BridgeLeave events.
331 AGI (Asterisk Gateway Interface)
333 * The manager event AGIExec has been split into AGIExecStart and AGIExecEnd.
335 * The manager event AsyncAGI has been split into AsyncAGIStart, AsyncAGIExec,
338 * The CONTROL STREAM FILE command now accepts an offsetms parameter. This
339 will start the playback of the audio at the position specified. It will
340 also return the final position of the file in 'endpos'.
342 * The SAY ALPHA command now accepts an additional parameter to control
343 whether it specifies the case of uppercase, lowercase, or all letters to
344 provide functionality similar to SayAlphaCase.
346 CDR (Call Detail Records)
348 * Significant changes have been made to the behavior of CDRs. For a full
349 definition of CDR behavior in Asterisk 12, please read the specification
350 on the Asterisk wiki (wiki.asterisk.org).
352 * CDRs will now be created between all participants in a bridge. For each
353 pair of channels in a bridge, a CDR is created to represent the path of
354 communication between those two endpoints. This lets an end user choose who
355 to bill for what during bridge operations with multiple parties.
357 * The duration, billsec, start, answer, and end times now reflect the times
358 associated with the current CDR for the channel, as opposed to a cumulative
359 measurement of all CDRs for that channel.
361 * When a CDR is dispatched, user defined CDR variables from both parties are
362 included in the resulting CDR. If both parties have the same variable, only
363 the Party A value is provided.
365 CEL (Channel Event Logging)
367 * The 'extra' field of all CEL events that use it now consists of a JSON blob
368 with key/value pairs which are defined in the Asterisk 12 CEL documentation.
370 * AST_CEL_BLINDTRANSFER events now report the transferee bridge unique
371 identifier, extension, and context in a JSON blob as the extra string
372 instead of the transferee channel name as the peer.
374 * AST_CEL_ATTENDEDTRANSFER events now report the peer as NULL and additional
375 information in the 'extra' string as a JSON blob. For transfers that occur
376 between two bridged channels, the 'extra' JSON blob contains the primary
377 bridge unique identifier, the secondary channel name, and the secondary
378 bridge unique identifier. For transfers that occur between a bridged channel
379 and a channel running an app, the 'extra' JSON blob contains the primary
380 bridge unique identifier, the secondary channel name, and the app name.
382 * AST_CEL_LOCAL_OPTIMIZE events have been added to convey local channel
383 optimizations with the record occurring for the semi-one channel and
384 the semi-two channel name in the peer field.
388 * The BRIDGE_FEATURES channel variable would previously only set features for
389 the calling party and would set this feature regardless of whether the
390 feature was in caps or in lowercase. Use of a caps feature for a letter
391 will now apply the feature to the calling party while use of a lowercase
392 letter will apply that feature to the called party.
394 * Add support for automixmon to the BRIDGE_FEATURES channel variable.
396 * Parking has been pulled from core and placed into a separate module called
397 res_parking. See Parking changes below for more details.
399 * You can now have the settings for a channel updated using the FEATURE()
400 and FEATUREMAP() functions inherited to child channels by setting
401 FEATURE(inherit)=yes.
403 * automixmon now supports additional channel variables from automon including:
404 TOUCH_MIXMONITOR_PREFIX, TOUCH_MIXMONITOR_MESSAGE_START,
405 and TOUCH_MIXMONITOR_MESSAGE_STOP
407 * A new general features.conf option 'recordingfailsound' has been added which
408 allowssetting a failure sound for a user tries to invoke a recording feature
409 such as automon or automixmon and it fails.
413 * When performing queue pause/unpause on an interface without specifying an
414 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
415 least one member of any queue exists for that interface.
417 * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
418 for realtime queue log entries.
422 * Parking is now implemented as a module instead of as core functionality.
423 The preferred way to configure parking is now through res_parking.conf while
424 configuration through features.conf is not currently supported.
426 * res_parking uses the configuration framework. If an invalid configuration is
427 supplied, res_parking will fail to load or fail to reload. Previously parking
428 lots that were misconfigured would generally be accepted with certain
429 configuration problems leading to individual disabled parking lots.
431 * Parked calls are now placed in bridges. This is a largely architectural change,
432 but it could have some implications in allowing for new parked call retrieval
433 methods and the contents of parking lots will be visible though certain bridge
436 * The order of arguments for the new parking applications are different from the
437 old ones to be more intuitive. Timeout and return context/exten/priority are now
438 implemented as options. parking_lot_name is now the first parameter. See the
439 application documentation for Park, ParkedCall, and ParkAndAnnounce for more
440 in-depth information as well as syntax.
442 * Extensions are no longer automatically created in the dialplan to park calls,
443 pickup parked calls, etc by default.
445 * adsipark is no longer supported under the new parking model
447 * The PARKINGSLOT channel variable has been deprecated in favor of PARKING_SPACE
448 to match the naming scheme of the new system.
450 * PARKING_SPACE and PARKEDLOT channel variables will now be set for a parked
451 channel even when comebactoorigin=yes
453 * New CLI command 'parking show' allows you to inspect the currently in use
454 parking lots. 'parking show <parkinglot>' will also show the parked calls
455 in that specific parking lot.
457 * The CLI command 'parkedcalls' is now deprecated in favor of
458 'parking show <parkinglot>'.
460 * The AMI command 'ParkedCalls' will now accept a 'ParkingLot' argument which
461 can be used to get a list of parked calls only for a specific parking lot.
463 * The AMI command 'Park' has had the argument 'Channel2' renamed to
464 'TimeoutChannel'. 'TimeoutChannel' is no longer a required argument.
465 'Channel2' can still be used as the argument name, but it is deprecated
466 and the 'TimeoutChannel' argument will be used if both are present.
468 * The ParkAndAnnounce application is now provided through res_parking instead
469 of through the separate app_parkandannounce module.
471 * ParkAndAnnounce will no longer go to the next position in dialplan on timeout
472 by default. Instead, it will follow the timeout rules of the parking lot. The
473 old behavior can be reproduced by using the 'c' option.
475 * Dynamic parking lots will now fail to be created if the parking lot specified
476 by PARKINGDYNAMIC does not exist.
478 * Dynamic parking lots will also fail to be created now if they require exclusive
479 park and parkedcall extensions which overlap with other parking lots.
481 * Dynamic parking lots will be cleared on reload for dynamic parking lots that
482 currently contain no calls. Dynamic parking lots containing parked calls will
483 persist through the reloads without alteration.
485 * If parkext_exclusive is set for a parking lot and that extension is already in
486 use when that parking lot tries to register it, this is now considered a parking
487 system configuration error. Configurations which do this will be rejected.
488 Dynamic parking lots which try to register extensions that already exist will
491 * Added a channel variable PARKER_FLAT which stores the name of the extension
492 that would be used to come back to if comebacktoorigin was set to use. This can
493 be useful when comebacktoorigin is off if you still want to use the extensions
494 in the park-dial context that are generated to redial the parker on timeout.
498 * Dynamic realtime tables for SIP Users can now include a 'path' field. This
499 will store the path information for that peer when it registers. Realtime
500 tables can also use the 'supportpath' field to enable Path header support.
502 * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
503 objectIdentifier. This maps to the supportpath option in sip.conf.
507 * All future modules which utilize Sorcery for object persistence must have a
508 column named "id" within their schema when using the Sorcery realtime module.
509 This column must be able to contain a string of up to 128 characters in length.
511 Security Events Framework
512 -------------------------
513 * Security Event timestamps now use ISO 8601 formatted date/time instead of the
514 "seconds-microseconds" format that it was using previously.
519 * When a channel driver is configured to enable jiterbuffers, they are now
520 applied unconditionally when a channel joins a bridge. If a jitterbuffer
521 is already set for that channel when it enters, such as by the JITTERBUFFER
522 function, then the existing jitterbuffer will be used and the one set by
523 the channel driver will not be applied.
527 * The updatecdr option has been removed. Altering the names of channels on a
528 CDR is not supported - the name of the channel is the name of the channel,
529 and pretending otherwise helps no one.
530 * The AGENTUPDATECDR channel variable has also been removed, for the same
531 reason as the updatecdr option.
532 * The driver is no longer a Data retrieval API data provider for the
534 * The endcall and enddtmf configuration options are removed. Use the
535 dialplan function CHANNEL(dtmf-features) to set DTMF features on the agent
536 channel before calling AgentLogin.
537 * chan_agent is removed and replaced with AgentLogin and AgentRequest dialplan
538 applications. Agents are connected with callers using the new AgentRequest
539 dialplan application. The Agents:<agent-id> device state is available to
540 monitor the status of an agent. See agents.conf.sample for valid
541 configuration options.
545 * chan_bridge is removed and its functionality is incorporated into ConfBridge
550 * The /b option is removed.
552 * chan_local moved into the system core and is no longer a loadable module.
556 * Added general support for busy detection.
558 * Added ECAM command support for Sony Ericsson phones.
562 * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
563 using the 'supportpath' setting, either on a global basis or on a peer basis.
564 This setting enables Asterisk to route outgoing out-of-dialog requests via a
565 set of proxies by using a pre-loaded route-set defined by the Path headers in
566 the REGISTER request. See Realtime updates for more configuration information.
568 * The SIP_CODEC family of variables may now specify more than one codec. Each
569 codec must be separated by a comma. The first codec specified is the
570 preferred codec for the offer. This allows a dialplan writer to specify both
571 audio and video codecs, e.g., Set(SIP_CODEC=ulaw,h264)
578 * JITTERBUFFER now accepts an argument of 'disabled' which can be used
579 to remove jitterbuffers previously set on a channel with JITTERBUFFER.
580 The value of this setting is ignored when disabled is used for the argument.
584 * The 'amaflags' and 'accountcode' attributes for the CDR function are
585 deprecated. Use the CHANNEL function instead to access these attributes.
586 * The 'l' option has been removed. When reading a CDR attribute, the most
587 recent record is always used. When writing a CDR attribute, all non-finalized
589 * The 'r' option has been removed, for the same reason as the 'l' option.
590 * The 's' option has been removed, as LOCKED semantics no longer exist in the
595 * A new function CDR_PROP has been added. This function lets you set properties
596 on a channel's active CDRs. This function is write-only. Properties accept
597 boolean values to set/clear them on the channel's CDRs. Valid properties
599 * 'party_a' - make this channel the preferred Party A in any CDR between two
600 channels. If two channels have this property set, the creation time of the
601 channel is used to determine who is Party A. Note that dialed channels are
602 never Party A in a CDR.
603 * 'disable' - disable CDRs on this channel. This is analogous to the NoCDR
604 application when set to True, and analogous to the 'e' option in ResetCDR
613 * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
614 them, an Asterisk-specific version of pjproject needs to be installed.
615 Tarballs are available from https://github.com/asterisk/pjproject/tags/.
619 * Device state for XMPP buddies is now available using the following format:
620 XMPP/<client name>/<buddy address>
621 If any resource is available the device state is considered to be not in use.
622 If no resources exist or all are unavailable the device state is considered
631 * The safe_asterisk script will now install over previously installations.
632 In previous versions of Asterisk, once installed a 'make install' would
633 skip over safe_asterisk if it was already installed.
634 * Certain options in safe_asterisk can now be configured from the
635 safe_asterisk.conf file. A sample version of this is located in the
638 ------------------------------------------------------------------------------
639 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
640 ------------------------------------------------------------------------------
646 * The Asterisk build system will now build and install a shared library
647 (libasteriskssl.so) used to wrap various initialization and shutdown functions
648 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
649 that Asterisk can ensure that these functions do *not* get called by any
650 modules that are loaded into Asterisk, since they should only be called once
651 in any single process. If desired, this feature can be disabled by supplying
652 the "--disable-asteriskssl" option to the configure script.
654 * A new make target, 'full', has been added to the Makefile. This performs
655 the same compilation actions as make all, but will also scan the entirety of
656 each source file for documentation. This option is needed to generate AMI
657 event documentation. Note that your system must have Python in order for
658 this make target to succeed.
660 * The optimization portion of the build system has been reworked to avoid
661 broken builds on certain architectures. All architecture-specific
662 optimization has been removed in favor of using -march=native to allow gcc
663 to detect the environment in which it is running when possible. This can
664 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
666 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
667 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
669 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
670 previously parsed the header file to obtain the version of Asterisk, you
671 will now have to go through Asterisk to get the version information.
679 * Added 'F()' option. Similar to the dial option, this can be supplied with
680 arguments indicating where the callee should go after the caller is hung up,
681 or without options specified, the priority after the Queue will be used.
686 * Added menu action admin_toggle_mute_participants. This will mute / unmute
687 all non-admin participants on a conference. The confbridge configuration
688 file also allows for the default sounds played to all conference users when
689 this occurs to be overriden using sound_participants_unmuted and
690 sound_participants_muted.
692 * Added menu action participant_count. This will playback the number of
693 current participants in a conference.
695 * Added announcement configuration option to user profile. If set the sound
696 file will be played to the user, and only the user, upon joining the
699 * Added record_file_append option that defaults to "yes", but if set to no
700 will create a new file between each start/stop recording.
705 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
706 channels respectively before the callee channels are called.
711 * Added support for IPv6.
713 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
714 external process will cause the current playlist to be cleared, including
715 stopping any audio file that is currently playing. This is useful when you
716 want to interrupt audio playback only when specific DTMF is entered by the
722 * A new option, 'I' has been added to app_followme. By setting this option,
723 Asterisk will not update the caller with connected line changes when they
724 occur. This is similar to app_dial and app_queue.
726 * The 'N' option is now ignored if the call is already answered.
728 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
729 and caller channels respectively before the callee channels are called.
731 * The winning FollowMe outgoing call is now put on hold if the caller put it on
737 * MixMonitor hooks now have IDs associated with them which can be used to
738 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
739 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
740 now accepts that ID as an argument.
742 * Added 'm' option, which stores a copy of the recording as a voicemail in the
748 * The connect action in app_mysql now allows you to specify a port number to
749 connect to. This is useful if you run a MySQL server on a non-standard
755 * Increased the default number of allowed destinations from 5 to 12.
760 * The app_page application now no longer depends on DAHDI or app_meetme. It
761 has been re-architected to use app_confbridge internally.
766 * Added queue options autopausebusy and autopauseunavail for automatically
767 pausing a queue member when their device reports busy or congestion.
769 * The 'ignorebusy' option for queue members has been deprecated in favor of
770 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
771 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
772 per interface basis. Individual ringinuse values can now be set in
773 queues.conf via an argument to member definitions. Lastly, the queue
774 'ringinuse' setting now only determines defaults for the per member
775 'ringinuse' setting and does not override per member settings like it does
778 * Added 'F()' option. Similar to the dial option, this can be supplied with
779 arguments indicating where the callee should go after the caller is hung up,
780 or without options specified, the priority after the Queue will be used.
782 * Added new option log_member_name_as_agent, which will cause the membername to
783 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
784 state_interface has been set.
786 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
788 * App_queue will now play periodic announcements for the caller that
789 holds the first position in the queue while waiting for answer.
793 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
794 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
795 changed arguments to SayUnixTime so that every option is truly optional even
796 when using multiple options (so that j option could be used without having to
797 manually specify timezone and format) There are other benefits, e.g., format
798 can now be used without specifying time zone as well.
803 * Addition of the VM_INFO function - see Function changes.
805 * The imapserver, imapport, and imapflags configuration options can now be
806 overriden on a user by user basis.
808 * When voicemail plays a message's envelope with saycid set to yes, when
809 reaching the caller id field it will play a recording of a file with the same
810 base name as the sender's callerid if there is a similarly named file in
811 <astspooldir>/recordings/callerids/
813 * Voicemails now contains a unique message identifier "msg_id", which is stored
814 in the message envelope with the sound files. IMAP backends will now store
815 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
816 backends will store the message identifier in a "msg_id" column. See
817 UPGRADE.txt for more information.
819 * Added VoiceMailPlayMsg application. This application will play a single
820 voicemail message from a mailbox. The result of the application, SUCCESS or
821 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
826 * Hangup handlers can be attached to channels using the CHANNEL() function.
827 Hangup handlers will run when the channel is hung up similar to the h
828 extension. The hangup_handler_push option will push a GoSub compatible
829 location in the dialplan onto the channel's hangup handler stack. The
830 hangup_handler_pop option will remove the last added location, and optionally
831 replace it with a new GoSub compatible location. The hangup_handler_wipe
832 option will remove all locations on the stack, and optionally add a new
835 * The expression parser now recognizes the ABS() absolute value function,
836 which will convert negative floating point values to positive values.
838 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
839 control of faxdetect.
841 * Addition of the VM_INFO function that can be used to retrieve voicemail
842 user information, such as the email address and full name.
843 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
846 * The REDIRECTING function now supports the redirecting original party id
849 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
850 lets you set some of the configuration options from the [general] section
851 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
852 the key sequence used to activate built-in features, such as blindxfer,
853 and automon. See the built-in documentation for details.
855 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
856 instead of simply the uri. This is the format that MessageSend() can use
857 in the from parameter for outgoing SIP messages.
859 * Added the PRESENCE_STATE function. This allows retrieving presence state
860 information from any presence state provider. It also allows setting
861 presence state information from a CustomPresence presence state provider.
862 See AMI/CLI changes for related commands.
864 * Added the AMI_CLIENT function to make manager account attributes available
865 to the dialplan. It currently supports returning the current number of
866 active sessions for a given account.
868 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
869 and the REDIRECTING functions.
877 * Added a manager event "LocalBridge" for local channel call bridges between
878 the two pseudo-channels created.
883 * Added dialtone_detect option for analog ports to disconnect incoming
884 calls when dialtone is detected.
886 * Added option colp_send to send ISDN connected line information. Allowed
887 settings are block, to not send any connected line information; connect, to
888 send connected line information on initial connect; and update, to send
889 information on any update during a call. Default is update.
891 * Add options namedcallgroup and namedpickupgroup to support installations
892 where a higher number of groups (>64) is required.
894 * Added support to use private party ID information with PRI calls.
899 * A new channel driver named chan_motif has been added which provides support for
900 Google Talk and Jingle in a single channel driver. This new channel driver includes
901 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
902 hold, unhold, and ringing notification. It is also compliant with the current Jingle
903 specification, current Google Jingle specification, and the original Google Talk
909 * Added NAT support for RTP. Setting in config is 'nat', which can be set
910 globally and overriden on a peer by peer basis.
912 * Direct media functionality has been added. Options in config are:
913 directmedia (directrtp) and directrtpsetup (earlydirect)
915 * ChannelUpdate events now contain a CallRef header.
920 * Asterisk will no longer substitute CID number for CID name in the display
921 name field if CID number exists without a CID name. This change improves
922 compatibility with certain device features such as Avaya IP500's directory
925 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
926 created using that setting to not be removed during SIP reload.
928 * Added settings recordonfeature and recordofffeature. When receiving an INFO
929 request with a "Record:" header, this will turn the requested feature on/off.
930 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
931 dynamic features must be enabled and configured properly on the requesting
932 channel for this to function properly.
934 * Add support to realtime for the 'callbackextension' option.
936 * When multiple peers exist with the same address, but differing
937 callbackextension options, incoming requests that are matched by address
938 will be matched to the peer with the matching callbackextension if it is
941 * Two new NAT options, auto_force_rport and auto_comedia, have been added
942 which set the force_rport and comedia options automatically if Asterisk
943 detects that an incoming SIP request crossed a NAT after being sent by
946 * The default global nat setting in sip.conf has been changed from force_rport
949 * NAT settings are now a combinable list of options. The equivalent of the
950 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
952 * Adds an option send_diversion which can be disabled to prevent
953 diversion headers from automatically being added to INVITE requests.
955 * Add support for lightweight NAT keepalive. If enabled a blank packet will
956 be sent to the remote host at a given interval to keep the NAT mapping open.
957 This can be enabled using the keepalive configuration option.
959 * Add option 'tonezone' to specify country code for indications. This option
960 can be set both globally and overridden for specific peers.
962 * The SIP Security Events Framework now supports IPv6.
964 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
965 between multiple user agents. When set, for directmedia reinvites,
966 Asterisk will not send an immediate reinvite on an incoming call leg. This
967 option is useful when peered with another SIP user agent that is known to
968 send immediate direct media reinvites upon call establishment.
970 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
973 * Add options subminexpiry and submaxexpiry to set limits of subscription
974 timer independently from registration timer settings. The setting of the
975 registration timer limits still is done by options minexpiry, maxexpiry
976 and defaultexpiry. For backwards compatibility the setting of minexpiry
977 and maxexpiry also is used to configure the subscription timer limits if
978 subminexpiry and submaxexpiry are not set in sip.conf.
980 * Set registration timer limits to default values when reloading sip
981 configuration and values are not set by configuration.
983 * Add options namedcallgroup and namedpickupgroup to support installations
984 where a higher number of groups (>64) is required.
986 * When a MESSAGE request is received, the address the request was received from
987 is now saved in the SIP_RECVADDR variable.
989 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
990 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
991 the ANI2/OLI information is set on the channel, which can be retrieved using
992 the CALLERID function.
994 * Peers can now be configured to support negotiation of ICE candidates using
995 the setting icesupport. See res_rtp_asterisk changes for more information.
997 * Added support for format attribute negotiation. See the Codecs changes for
1000 * Extra headers specified with SIPAddHeader are sent with the REFER message
1001 when using Transfer application. See refer_addheaders in sip.conf.sample.
1003 * Added support to use private party ID information with calls.
1005 * Adds an option discard_remote_hold_retrieval that when set stops telling
1006 the peer to start music on hold.
1011 * Added skinny version 17 protocol support.
1015 --------------------
1016 * Added ability to use multiple lines for a single phone. This allows multiple
1017 calls to occur on a single phone, using callwaiting and switching between calls.
1019 * Added option 'sharpdial' allowing end dialing by pressing # key
1021 * Added option 'interdigit_timer' to control phone dial timeout
1023 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
1025 * Added global 'debug' option, that enables debug in channel driver
1027 * Added ability to translate on-screen menu in multiple languages. Tested on
1028 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
1029 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
1032 * In addition to English added French and Russian languages for on-screen menus
1034 * Reworked dialing number input: added dialing by timeout, immediate dial on
1035 on dialplan compare, phone number length now not limited by screen size
1037 * Added ability to pickup a call using features.conf defined value and
1043 * Add options namedcallgroup and namedpickupgroup to support installations
1044 where a higher number of groups (>64) is required.
1046 * Added support to use private party ID information with calls.
1051 * The minimum DTMF duration can now be configured in asterisk.conf
1052 as "mindtmfduration". The default value is (as before) set to 80 ms.
1053 (previously it was only available in source code)
1055 * Named ACLs can now be specified in acl.conf and used in configurations that
1056 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
1057 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
1058 working ACL. In addition, some CLI commands have been added to provide
1059 show information and allow for module reloading - see CLI Changes.
1061 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
1062 items (separated by commas), and items in the rule can be negated by prefixing
1063 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
1064 longer necessray to control the order that the 'permit' and 'deny' columns are
1065 returned from queries.
1067 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
1068 be used within the dynamic weight attribute when specifying a mapping.
1070 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
1071 header, instead of putting the user defined event name there. When enabled
1072 the UserDefType header is added for user defined events. This feature is
1073 enabled with the setting show_user_defined.
1075 * Macro has been deprecated in favor of GoSub. For redirecting and connected
1076 line purposes use the following variables instead of their macro equivalents:
1077 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
1078 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
1079 cc_callback_macro in channel configurations.
1081 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
1084 * Call files now support the "early_media" option to connect with an outgoing
1085 extension when early media is received.
1087 * Added support to use private party ID information with calls.
1092 * A new channel variable, AGIEXITONHANGUP, has been added which allows
1093 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
1094 AGI application would exit immediately after a channel hangup is detected.
1096 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
1097 are resolved and each address is attempted in turn until one succeeds or
1101 AMI (Asterisk Manager Interface)
1103 * The originate action now has an option "EarlyMedia" that enables the
1104 call to bridge when we get early media in the call. Previously,
1105 early media was disregarded always when originating calls using AMI.
1107 * Added setvar= option to manager accounts (much like sip.conf)
1109 * Originate now generates an error response if the extension given is not found
1112 * MixMonitor will now show IDs associated with the mixmonitor upon creating
1113 them if the i(variable) option is used. StopMixMonitor will accept
1114 MixMonitorID as an option to close specific MixMonitors.
1116 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
1117 updated to include information about peers configured with
1118 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
1119 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
1120 returned if auto_force_rport is not enabled.
1122 * Added SIPpeerstatus manager command which will generate PeerStatus events
1123 similar to the existing PeerStatus events found in chan_sip on demand.
1125 * Hangup now can take a regular expression as the Channel option. If you want
1126 to hangup multiple channels, use /regex/ as the Channel option. Existing
1127 behavior to hanging up a single channel is unchanged, but if you pass a regex,
1128 the manager will send you a list of channels back that were hung up.
1130 * Support for IPv6 addresses has been added.
1132 * AMI Events can now be documented in the Asterisk source. Note that AMI event
1133 documentation is only generated when Asterisk is compiled using 'make full'.
1134 See the CLI section for commands to display AMI event information.
1136 * The AMI Hangup event now includes the AccountCode header so you can easily
1137 correlate with AMI Newchannel events.
1139 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
1140 the StateInterface of the queue member.
1142 * Added AMI event SessionTimeout in the Call category that is issued when a
1143 call is terminated due to either RTP stream inactivity or SIP session timer
1146 * CEL events can now contain a user defined header UserDefType. See core
1147 changes for more information.
1149 * OOH323 ChannelUpdate events now contain a CallRef header.
1151 * Added PresenceState command. This command will report the presence state for
1152 the given presence provider.
1154 * Added Parkinglots command. This will list all parking lots as a series of
1155 AMI Parkinglot events.
1157 * Added MessageSend command. This behaves in the same manner as the
1158 MessageSend application, and is a technolgoy agnostic mechanism to send out
1159 of call text messages.
1161 * Added "message" class authorization. This grants an account permission to
1162 send out of call messages. Write-only.
1167 * The "dialplan add include" command has been modified to create context a context
1168 if one does not already exist. For instance, "dialplan add include foo into bar"
1169 will create context "bar" if it does not already exist.
1171 * A "dialplan remove context" command has been added to remove a context from
1174 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
1175 filenames of all running mixmonitors on a channel.
1177 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
1178 numeric instead of 0, 1, or 2.
1180 * "stun show status" will show a table describing how the STUN client is
1183 * "acl show [named acl]" will show information regarding a Named ACL. The
1184 acl module can be reloaded with "reload acl".
1186 * Added CLI command to display AMI event information - "manager show events",
1187 which shows a list of all known and documented AMI events, and "manager show
1188 event [event name]", which shows detail information about a specific AMI
1191 * The result of the CLI command "queue show" now includes the state interface
1192 information of the queue member.
1194 * The command "core set verbose" will now set a separate level of logging for
1195 each remote console without affecting any other console.
1197 * Added command "cdr show pgsql status" to check connection status
1199 * "sip show channel" will now display the complete route set.
1201 * Added "presencestate list" command. This command will list all custom
1202 presence states that have been set by using the PRESENCE_STATE dialplan
1205 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
1206 command. This changes a custom presence to a new state.
1211 * Codec lists may now be modified by the '!' character, to allow succinct
1212 specification of a list of codecs allowed and disallowed, without the
1213 requirement to use two different keywords. For example, to specify all
1214 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
1216 * Add support for parsing SDP attributes, generating SDP attributes, and
1217 passing it through. This support includes codecs such as H.263, H.264, SILK,
1218 and CELT. You are able to set up a call and have attribute information pass.
1219 This should help considerably with video calls.
1221 * The iLBC codec can now use a system-provided iLBC library if one is installed,
1222 just like the GSM codec.
1226 * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
1227 'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
1231 * Asterisk version and build information is now logged at the beginning of a
1234 * Threads belonging to a particular call are now linked with callids which get
1235 added to any log messages produced by those threads. Log messages can now be
1236 easily identified as involved with a certain call by looking at their call id.
1237 Call ids may also be attached to log messages for just about any case where
1238 it can be determined to be related to a particular call.
1240 * Each logging destination and console now have an independent notion of the
1241 current verbosity level. Logger.conf now allows an optional argument to
1242 the 'verbose' specifier, indicating the level of verbosity sent to that
1243 particular logging destination. Additionally, remote consoles now each
1244 have their own verbosity level. The command 'core set verbose' will now set
1245 a separate level for each remote console without affecting any other
1251 * Added 'announcement' option which will play at the start of MOH and between
1252 songs in modes of MOH that can detect transitions between songs (eg.
1258 * New per parking lot options: comebackcontext and comebackdialtime. See
1259 configs/features.conf.sample for more details.
1261 * Channel variable PARKER is now set when comebacktoorigin is disabled in
1264 * Channel variable PARKEDCALL is now set with the name of the parking lot
1265 when a timeout occurs.
1271 CDR Postgresql Driver
1273 * Added command "cdr show pgsql status" to check connection status
1276 CDR Adaptive ODBC Driver
1278 * Added schema option for databases that support specifying a schema.
1286 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
1287 CALENDAR_WRITE has completed successfully.
1292 * A new option, 'probation' has been added to rtp.conf
1293 RTP in strictrtp mode can now require more than 1 packet to exit learning
1294 mode with a new source (and by default requires 4). The probation option
1295 allows the user to change the required number of packets in sequence to any
1296 desired value. Use a value of 1 to essentially restore the old behavior.
1297 Also, with strictrtp on, Asterisk will now drop all packets until learning
1298 mode has successfully exited. These changes are based on how pjmedia handles
1299 media sources and source changes.
1301 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
1302 enabled or disabled using the icesupport setting. A variety of other
1303 settings have been introduced to configure STUN/TURN connections.
1308 * A new module, res_corosync, has been introduced. This module uses the
1309 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
1310 of Asterisk servers to both Message Waiting Indication (MWI) and/or
1311 Device State (presence) information. This module is very similar to, and
1312 is a replacement for the res_ais module that was in previous releases of
1318 * This module adds a cleaned up, drop-in replacement for res_jabber called
1319 res_xmpp. This provides the same externally facing functionality but is
1320 implemented differently internally. res_jabber has been deprecated in favor
1321 of res_xmpp; please see the UPGRADE.txt file for more information.
1326 * The safe_asterisk script has been updated to allow several of its parameters
1327 to be set from environment variables. This also enables a custom run
1328 directory of Asterisk to be specified, instead of defaulting to /tmp.
1330 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
1331 its value to determine the directory to assume is the top-level directory of
1332 the source tree. If the variable is not set, it defaults to the current
1333 behavior and uses the current working directory.
1335 ------------------------------------------------------------------------------
1336 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
1337 ------------------------------------------------------------------------------
1341 * Asterisk now has protocol independent support for processing text messages
1342 outside of a call. Messages are routed through the Asterisk dialplan.
1343 SIP MESSAGE and XMPP are currently supported. There are options in
1344 jabber.conf and sip.conf to allow enabling these features.
1345 -> jabber.conf: see the "sendtodialplan" and "context" options.
1346 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
1347 and "outofcall_message_context" options.
1348 The MESSAGE() dialplan function and MessageSend() application have been
1349 added to go along with this functionality. More detailed usage information
1350 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
1351 * If real-time text support (T.140) is negotiated, it will be preferred for
1352 sending text via the SendText application. For example, via SIP, messages
1353 that were once sent via the SIP MESSAGE request would be sent via RTP if
1354 T.140 text is negotiated for a call.
1358 * parkedmusicclass can now be set for non-default parking lots.
1360 Asterisk Manager Interface
1361 --------------------------
1362 * PeerStatus now includes Address and Port.
1363 * Added Hold events for when the remote party puts the call on and off hold
1364 for chan_dahdi ISDN channels.
1365 * Added new action MeetmeListRooms to list active conferences (shows same
1366 data as "meetme list" at the CLI).
1367 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
1368 Description field that is set by 'description' in the channel configuration
1370 * Added Uniqueid header to UserEvent.
1371 * Added new action FilterAdd to control event filters for the current session.
1372 This requires the system permission and uses the same filter syntax as
1373 filters that can be defined in manager.conf
1374 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
1375 versions had some instances of the event converted, but others were left
1376 as-is. All Unlink events should now be converted to Bridge events. The AMI
1377 protocol version number was incremented to 1.2 as a result of this change.
1379 Asterisk HTTP Server
1380 --------------------------
1381 * The HTTP Server can bind to IPv6 addresses.
1384 --------------------------
1385 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
1386 with busydetect. usage example: busypattern=200,200,200,600
1389 --------------------------
1390 * New 'gtalk show settings' command showing the current settings loaded from
1392 * The 'logger reload' command now supports an optional argument, specifying an
1393 alternate configuration file to use.
1394 * 'dialplan add extension' command will now automatically create a context if
1395 the specified context does not exist with a message indicated it did so.
1396 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
1397 Description field which can be populated with 'description' in the channel
1398 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
1401 --------------------------
1402 * The filter option in cdr_adaptive_odbc now supports negating the argument,
1403 thus allowing records which do NOT match the specified filter.
1404 * Added ability to log CONGESTION calls to CDR
1407 --------------------------
1408 * Ability to define custom SILK formats in codecs.conf.
1409 * Addition of speex32 audio format with translation.
1410 * CELT codec pass-through support and ability to define
1411 custom CELT formats in codecs.conf.
1412 * Ability to read raw signed linear files with sample rates
1413 ranging from 8khz - 192khz. The new file extensions introduced
1414 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
1415 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
1416 Skinny, H.323, etc) can still only support the following codecs:
1417 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
1418 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
1419 Video: h261, h263, h263p, h264, mpeg4
1424 --------------------------
1425 * New highly optimized and customizable ConfBridge application capable of
1426 mixing audio at sample rates ranging from 8khz-96khz.
1427 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
1428 and bridge profiles on a channel.
1429 * CONFBRIDGE_INFO dialplan function capable of retrieving information
1430 about a conference such as locked status and number of parties, admins,
1432 * Addition of video_mode option in confbridge.conf for adding video support
1433 into a bridge profile.
1434 * Addition of the follow_talker video_mode in confbridge.conf. This video
1435 mode dynamically switches the video feed to always display the loudest talker
1436 supplying video in the conference.
1440 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
1441 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
1442 variables from asterisk.conf.
1446 * Addition of the JITTERBUFFER dialplan function. This function allows
1447 for jitterbuffering to occur on the read side of a channel. By using
1448 this function conference applications such as ConfBridge and MeetMe can
1449 have the rx streams jitterbuffered before conference mixing occurs.
1450 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
1452 * Added STRREPLACE function. This function let's the user search a variable
1453 for a given string to replace with another string as many times as the
1454 user specifies or just throughout the whole string.
1455 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
1456 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
1457 * Added extensions to chan_ooh323 in function CHANNEL()
1459 libpri channel driver (chan_dahdi) DAHDI changes
1460 --------------------------
1461 * Added moh_signaling option to specify what to do when the channel's bridged
1462 peer puts the ISDN channel on hold.
1463 * Added display_send and display_receive options to control how the display ie
1464 is handled. To send display text from the dialplan use the SendText()
1465 application when the option is enabled.
1466 * Added mcid_send option to allow sending a MCID request on a span.
1469 --------------------------
1470 * Added setvar option to calendar.conf to allow setting channel variables on
1471 notification channels.
1472 * Added "calendar show types" CLI command to list registered calendar
1476 --------------------------
1477 * Added two new options, r and t with file name arguments to record
1478 single direction (unmixed) audio recording separate from the bidirectional
1479 (mixed) recording. The mixed file name argument is optional now as long
1480 as at least one recording option is used.
1483 --------------------------
1484 * Added a new option, l, which will disable local call optimization for
1485 channels involved with the FollowMe thread. Use this option to improve
1486 compatability for a FollowMe call with certain dialplan apps, options, and
1490 --------------------------
1491 * Added option "k" that will automatically close the conference when there's
1492 only one person left when a user exits the conference.
1495 --------------------------
1496 * cel_pgsql now supports the 'extra' column for data added using the
1497 CELGenUserEvent() application.
1500 --------------------------
1501 * Support for defining hints has been added to pbx_lua. See the 'hints' table
1502 in the sample extensions.lua file for syntax details.
1503 * Applications that perform jumps in the dialplan such as Goto will now
1504 execute properly. When pbx_lua detects that the context, extension, or
1505 priority we are executing on has changed it will immediately return control
1506 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
1507 the priority after the currently executing priority.
1508 * An autoservice is now started by default for pbx_lua channels. It can be
1509 stopped and restarted using the autoservice_stop() and autoservice_start()
1513 --------------------------
1514 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
1515 into a FAXStatus event with an 'Operation' header that will be either
1516 'send', 'receive', and 'gateway'.
1517 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
1518 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
1519 feature will handle converting a fax call between an audio T.30 fax terminal
1520 and an IFP T.38 fax terminal.
1524 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
1525 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
1526 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
1530 * Added general option negative_penalty_invalid default off. when set
1531 members are seen as invalid/logged out when there penalty is negative.
1532 for realtime members when set remove from queue will set penalty to -1.
1533 * Added queue option autopausedelay when autopause is enabled it will be
1534 delayed for this number of seconds since last successful call if there
1535 was no prior call the agent will be autopaused immediately.
1536 * Added member option ignorebusy this when set and ringinuse is not
1537 will allow per member control of multiple calls as ringinuse does for
1542 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
1544 * Added 'k' option to MeetMe to automatically kill the conference when there's only
1545 one participant left (much like a normal call bridge)
1546 * Added extra argument to Originate to set timeout.
1550 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
1551 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
1552 utility in the UTILS section of menuselect. If an existing astdb is found and no
1553 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
1554 convert an existing astdb to the SQLite3 version automatically at runtime.
1558 * Modules marked as deprecated are no longer marked as building by default. Enabling
1559 these modules is still available via menuselect.
1563 * authdebug is now disabled by default. To enable this functionaility again
1564 set authdebug = yes in iax.conf.
1568 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
1569 releases it was disabled.
1573 * The PBX core previously made a call with a non-existing extension test for
1574 extension s@default and jump there if the extension existed.
1575 This was a bad default behaviour and violated the principle of least surprise.
1576 It has therefore been changed in this release. It may affect some
1577 applications and configurations that rely on this behaviour. Most channel
1578 drivers have avoided this for many releases by testing whether the extension
1579 called exists before starting the PBX and generating a local error.
1580 This behaviour still exists and works as before.
1582 Extension "s" is used when no extension is given in a channel driver,
1583 like immediate answer in DAHDI or calling to a domain with no user part
1586 ------------------------------------------------------------------------------
1587 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
1588 ------------------------------------------------------------------------------
1592 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
1593 now defaults to force_rport. It is very important that phones requiring nat=no be
1594 specifically set as such instead of relying on the default setting. If at all
1595 possible, all devices should have nat settings configured in the general section as
1596 opposed to configuring nat per-device.
1597 * Added preferred_codec_only option in sip.conf. This feature limits the joint
1598 codecs sent in response to an INVITE to the single most preferred codec.
1599 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
1600 to be used for the outgoing call. It must be one of the codecs configured
1602 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
1603 to be used for holding a private key. If tlsprivatekey is not specified,
1604 tlscertfile is searched for both public and private key.
1605 * Added tlsclientmethod option to sip.conf. This allows the protocol for
1606 outbound client connections to be specified.
1607 * The sendrpid parameter has been expanded to include the options
1608 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
1609 header to be sent (equivalent to setting sendrpid=yes) and setting
1610 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
1611 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
1612 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
1613 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
1614 will accept the SDP even if the SDP version number is not properly incremented,
1615 but will generate a warning in the log indicating that the SIP peer that sent
1616 the SDP should have the 'ignoresdpversion' option set.
1617 * The 'nat' option has now been been changed to have yes, no, force_rport, and
1618 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
1619 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
1620 remote side requests it and disables symmetric RTP support. Setting it to
1621 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
1622 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
1623 and enables symmetric RTP support.
1624 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
1625 response. This permits the master channel to know how each channel dialled
1626 in a multi-channel setup resolved in an individual way. This carries a
1627 performance penalty and can be disabled in sip.conf using the
1628 'storesipcause' option.
1629 * Added 'externtcpport' and 'externtlsport' options to allow custom port
1630 configuration for the externip and externhost options when tcp or tls is used.
1631 * Added support for message body (stored in content variable) to SIP NOTIFY message
1632 accessible via AMI and CLI.
1633 * Added 'media_address' configuration option which can be used to explicitly specify
1634 the IP address to use in the SDP for media (audio, video, and text) streams.
1635 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
1636 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
1638 * Added 'use_q850_reason' configuration option for generating and parsing
1639 if available Reason: Q.850;cause=<cause code> header. It is implemented
1640 in some gateways for better passing PRI/SS7 cause codes via SIP.
1641 * When dialing SIP peers, a new component may be added to the end of the dialstring
1642 to indicate that a specific remote IP address or host should be used when dialing
1643 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
1644 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
1645 ability to selectively force bridged channels to also be encrypted is also
1646 implemented. Branching in the dialplan can be done based on whether or not
1647 a channel has secure media and/or signaling.
1648 * Added directmediapermit/directmediadeny to limit which peers can send direct media
1650 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
1651 Charge messages to snom phones.
1652 * Added support for G.719 media streams.
1653 * Added support for 16khz signed linear media streams.
1654 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
1655 RTP has been outfitted with the same abilities.
1656 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
1657 available in device configurations as well as in the dial plan.
1658 * Addition of the 'subscribe_network_change' option for turning on and off
1659 res_stun_monitor module support in chan_sip.
1660 * Addition of the 'auth_options_requests' option for turning on and off
1661 authentication for OPTIONS requests in chan_sip.
1665 * Add #tryinclude statement for config files. This provides the same
1666 functionality as the #include statement however an asterisk module will
1667 still load if the filename does not exist. Using the #include statement
1668 Asterisk will not allow the module to load.
1672 * Added rtsavesysname option into iax.conf to allow the systname to be saved
1673 on realtime updates.
1674 * Added the ability for chan_iax2 to inform the dialplan whether or not
1675 encryption is being used. This interoperates with the SIP SRTP implementation
1676 so that a secure SIP call can be bridged to a secure IAX call when the
1677 dialplan requires bridged channels to be "secure".
1678 * Addition of the 'subscribe_network_change' option for turning on and off
1679 res_stun_monitor module support in chan_iax.
1684 * Added ability to preset channel variables on indicated lines with the setvar
1685 configuration option. Also, clearvars=all resets the list of variables back
1687 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
1688 See configs/res_pktccops.conf for more information.
1690 XMPP Google Talk/Jingle changes
1691 -------------------------------
1692 * Added the externip option to gtalk.conf.
1693 * Added the stunaddr option to gtalk.conf which allows for the automatic
1694 retrieval of the external ip from a stun server.
1698 * Added 'p' option to PickupChan() to allow for picking up channel by the first
1699 match to a partial channel name.
1700 * Added .m3u support for Mp3Player application.
1701 * Added progress option to the app_dial D() option. When progress DTMF is
1702 present, those values are sent immediately upon receiving a PROGRESS message
1703 regardless if the call has been answered or not.
1704 * Added functionality to the app_dial F() option to continue with execution
1705 at the current location when no parameters are provided.
1706 * Added the 'a' option to app_dial to answer the calling channel before any
1707 announcements or macros are executed.
1708 * Modified app_dial to set answertime when the called channel answers even if
1709 the called channel hangs up during playback of an announcement.
1710 * Modified app_dial 'r' option to support an additional parameter to play an
1711 indication tone from indications.conf
1712 * Added c() option to app_chanspy. This option allows custom DTMF to be set
1713 to cycle through the next available channel. By default this is still '*'.
1714 * Added x() option to app_chanspy. This option allows DTMF to be set to
1715 exit the application.
1716 * The Voicemail application has been improved to automatically ignore messages
1717 that only contain silence.
1718 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
1719 associated mailbox(es) to be greetings-only.
1720 * The ChanSpy application now has the 'S' option, which makes the application
1721 automatically exit once it hits a point where no more channels are available
1723 * The ChanSpy application also now has the 'E' option, which spies on a single
1724 channel and exits when that channel hangs up.
1725 * The MeetMe application now turns on the DENOISE() function by default, for
1726 each participant. In our tests, this has significantly decreased background
1727 noise (especially noisy data centers).
1728 * Voicemail now permits storage of secrets in a separate file, located in the
1729 spool directory of each individual user. The control for this is located in
1730 the "passwordlocation" option in voicemail.conf. Please see the sample
1731 configuration for more information.
1732 * The ChanIsAvail application now exposes the returned cause code using a separate
1733 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
1734 * Added 'd' option to app_followme. This option disables the "Please hold"
1736 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
1737 received will terminate recording.
1738 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
1739 Previously the folder could only be set per context, but has now been extended
1740 using the imapfolder option.
1741 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
1742 * Voicemail now allows the pager date format to be specified separately from the
1744 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
1745 to allow joining, leaving, and sending text to group chats.
1746 * MeetMe has a new option 'G' to play an announcement before joining a conference.
1747 * Page has a new option 'A(x)' which will playback an announcement simultaneously
1748 to all paged phones (and optionally excluding the caller's one using the new
1749 option 'n') before the call is bridged.
1750 * The 'f' option to Dial has been augmented to take an optional argument. If no
1751 argument is provided, the 'f' option works as it always has. If an argument is
1752 provided, then the connected party information of all outgoing channels created
1753 during the Dial will be set to the argument passed to the 'f' option.
1754 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
1756 * The OSP lookup application adds in/outbound network ID, optional security,
1757 number portability, QoS reporting, destination IP port, custom info and service
1759 * Added new application VMSayName that will play the recorded name of the voicemail
1760 user if it exists, otherwise will play the mailbox number.
1761 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
1762 retrieve state for a particular bridge, where <name> is the conference name
1763 * app_directory now allows exiting at any time using the operator or pound key.
1764 * Voicemail now supports setting a locale per-mailbox.
1765 * Two new applications are provided for declining counting phrases in multiple
1766 languages. See the application notes for SayCountedNoun and SayCountedAdj for
1768 * Voicemail now runs the externnotify script when pollmailboxes is activated and
1770 * Voicemail now includes rdnis within msgXXXX.txt file.
1771 * ExternalIVR now supports IPv6 addresses.
1772 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
1773 at https://wiki.asterisk.org/wiki/x/oQBB
1774 * ParkedCall and Park can now specify the parking lot to use.
1778 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
1779 over SRV records associated with a specific service. From the CLI, type
1780 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
1781 details on how these may be used.
1782 * PITCH_SHIFT dialplan function added. This function can be used to modify the
1783 pitch of a channel's tx and rx audio streams.
1784 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
1785 setting various connected line and redirecting party information.
1786 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
1787 support ISDN subaddressing.
1788 * The CHANNEL() function now supports the "name" and "checkhangup" options.
1789 * For DAHDI channels, the CHANNEL() dialplan function now allows
1790 the dialplan to request changes in the configuration of the active
1791 echo canceller on the channel (if any), for the current call only.
1794 exten => s,n,Set(CHANNEL(echocan_mode)=off)
1796 The possible values are:
1798 on - normal mode (the echo canceller is actually reinitialized)
1800 fax - FAX/data mode (NLP disabled if possible, otherwise completely
1802 voice - voice mode (returns from FAX mode, reverting the changes that
1803 were made when FAX mode was requested)
1804 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
1805 and setting variables on the channel which created the current channel.
1806 Administrators should take care to avoid naming conflicts, when multiple
1807 channels are dialled at once, especially when used with the Local channel
1808 construct (which all could set variables on the master channel). Usage
1809 of the HASH() dialplan function, with the key set to the name of the slave
1810 channel, is one approach that will avoid conflicts.
1811 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
1813 * func_odbc now allows multiple row results to be retrieved without using
1814 mode=multirow. If rowlimit is set, then additional rows may be retrieved
1815 from the same query by using the name of the function which retrieved the
1816 first row as an argument to ODBC_FETCH().
1817 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
1818 dialplan. This function returns the content of the received message.
1819 * Added REPLACE, which searches a given variable name for a set of characters,
1820 then either replaces them with a single character or deletes them.
1821 * Added PASSTHRU, which literally passes the same argument back as its return
1822 value. The intent is to be able to use a literal string argument to
1823 functions that currently require a variable name as an argument.
1824 * HASH-associated variables now can be inherited across channel creation, by
1825 prefixing the name of the hash at assignment with the appropriate number of
1826 underscores, just like variables.
1827 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
1828 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
1829 whether or not channels that are bridged to the current channel will be
1830 required to have secure signaling and/or media.
1831 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
1832 the current channel has secure signaling and/or media.
1833 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
1834 "no_media_path" option.
1835 Returns "0" if there is a B channel associated with the call.
1836 Returns "1" if no B channel is associated with the call. The call is either
1837 on hold or is a call waiting call.
1838 * Added option to dialplan function CDR(), the 'f' option
1839 allows for high resolution times for billsec and duration fields.
1840 * FILE() now supports line-mode and writing.
1841 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
1842 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
1846 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
1847 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
1848 and is set when a dynamic feature is triggered.
1849 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
1850 to dynamically create a new parking lot matching the value this varible is
1852 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
1853 features.conf that should be the base for dynamic parkinglots.
1854 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
1855 parkinglot should have.
1856 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
1857 parkinglot should have.
1858 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
1863 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
1864 timeout has expired.
1865 * Added 'R' option to app_queue. This option stops moh and indicates ringing
1866 to the caller when an Agent's phone is ringing. This can be used to indicate
1867 to the caller that their call is about to be picked up, which is nice when
1868 one has been on hold for an extened period of time.
1869 * A new config option, penaltymemberslimit, has been added to queues.conf.
1870 When set this option will disregard penalty settings when a queue has too
1872 * A new option, 'I' has been added to both app_queue and app_dial.
1873 By setting this option, Asterisk will not update the caller with
1874 connected line changes or redirecting party changes when they occur.
1875 * A 'relative-periodic-announce' option has been added to queues.conf. When
1876 enabled, this option will cause periodic announce times to be calculated
1877 from the end of announcements rather than from the beginning.
1878 * The autopause option in queues.conf can be passed a new value, "all." The
1879 result is that if a member becomes auto-paused, he will be paused in all
1880 queues for which he is a member, not just the queue that failed to reach
1882 * Added dialplan function QUEUE_EXISTS to check if a queue exists
1883 * The queue logger now allows events to optionally propagate to a file,
1884 even when realtime logging is turned on. Additionally, realtime logging
1885 supports sending the event arguments to 5 individual fields, although it
1886 will fallback to the previous data definition, if the new table layout is
1889 mISDN channel driver (chan_misdn) changes
1890 ----------------------------------------
1891 * Added display_connected parameter to misdn.conf to put a display string
1892 in the CONNECT message containing the connected name and/or number if
1893 the presentation setting permits it.
1894 * Added display_setup parameter to misdn.conf to put a display string
1895 in the SETUP message containing the caller name and/or number if the
1896 presentation setting permits it.
1897 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
1898 indicate the dialplan settings are to be obtained from the asterisk
1900 * Made misdn.conf parameter callerid accept the "name" <number> format
1901 used by the rest of the system.
1902 * Made use the nationalprefix and internationalprefix misdn.conf
1903 parameters to prefix any received number from the ISDN link if that
1904 number has the corresponding Type-Of-Number. NOTE: This includes
1905 comparing the incoming call's dialed number against the MSN list.
1906 * Added the following new parameters: unknownprefix, netspecificprefix,
1907 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
1908 received number from the ISDN link if that number has the corresponding
1910 * Added new dialplan application misdn_command which permits controlling
1911 the CCBS/CCNR functionality.
1912 * Added new dialplan function mISDN_CC which permits retrieval of various
1913 values from an active call completion record.
1914 * For PTP, you should manually send the COLR of the redirected-to party
1915 for an incomming redirected call if the incoming call could experience
1916 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
1917 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
1918 if the REDIRECTING(from-num) is not empty.
1919 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
1920 option on all of the REDIRECTING statements before dialing the
1921 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
1922 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
1923 redirecting-to presentation (COLR) when it becomes available.
1924 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
1927 thirdparty mISDN enhancements
1928 -----------------------------
1929 mISDN has been modified by Digium, Inc. to greatly expand facility message
1931 * Enhanced COLP support for call diversion and transfer.
1932 * CCBS/CCNR support.
1934 The latest modified mISDN v1.1.x based version is available at:
1935 http://svn.digium.com/svn/thirdparty/mISDN/trunk
1936 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
1938 Tagged versions of the modified mISDN code are available under:
1939 http://svn.digium.com/svn/thirdparty/mISDN/tags
1940 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
1942 libpri channel driver (chan_dahdi) DAHDI changes
1943 -------------------------------------------
1944 * The channel variable PRIREDIRECTREASON is now just a status variable
1945 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
1946 to read and alter the reason.
1947 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
1948 redirected-to party for an incomming redirected call if the incoming call
1949 could experience further redirects. Just set the
1950 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
1951 to the COLR. A call has been redirected if the REDIRECTING(count) is not
1953 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
1954 use the inhibit(i) option on all of the REDIRECTING statements before
1955 dialing the redirected-to party. You still have to set the
1956 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
1957 will update the redirecting-to presentation (COLR) when it becomes available.
1958 * Added the ability to ignore calls that are not in a Multiple Subscriber
1959 Number (MSN) list for PTMP CPE interfaces.
1960 * Added dynamic range compression support for dahdi channels. It is
1961 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
1962 * Added support for ISDN calling and called subaddress with partial support
1963 for connected line subaddress.
1964 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
1965 * Added handling of received HOLD/RETRIEVE messages and the optional ability
1966 to transfer a held call on disconnect similar to an analog phone.
1967 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
1968 Will reroute/deflect an outgoing call when receive the message.
1969 Can use the DAHDISendCallreroutingFacility to send the message for the
1971 * Added standard location to add options to chan_dahdi dialing:
1972 Dial(DAHDI/g1[/extension[/options]])
1975 R Reverse charging indication
1976 * Added Reverse Charging Indication (Collect calls) send/receive option.
1977 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
1978 Dial(DAHDI/g1/extension/R)
1979 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
1980 (requires latest LibPRI)
1981 * Added ability to send/receive keypad digits in the SETUP message.
1982 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
1983 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
1984 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
1985 (requires latest LibPRI)
1986 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
1987 to eliminate tromboned calls. A tromboned call goes out an interface and comes
1988 back into the same interface. Tromboned calls happen because of call routing,
1989 call deflection, call forwarding, and call transfer.
1990 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
1991 * Added the ability to support call waiting calls. (The SETUP has no B channel
1993 * Added Malicious Call ID (MCID) event to the AMI call event class.
1994 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
1996 Asterisk Manager Interface
1997 --------------------------
1998 * The Hangup action now accepts a Cause header which may be used to
1999 set the channel's hangup cause.
2000 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
2001 to specify a separate .pem file to hold a private key. By default sslcert
2002 is used to hold both the public and private key.
2003 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
2004 for options containing the 'tls' prefix. For example, 'sslenable' is now
2005 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
2006 across all .conf files. All affected sample.conf files have been modified to
2007 reflect this change. Previous options such as 'sslenable' still work,
2008 but options with the 'tls' prefix are preferred.
2009 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
2010 in a channel. (res_mutestream.so)
2011 * The configuration file manager.conf now supports a channelvars option, which
2012 specifies a list of channel variables to include in each channel-oriented
2014 * The redirect command now has new parameters ExtraContext, ExtraExtension,
2015 and ExtraPriority to allow redirecting the second channel to a different
2016 location than the first.
2017 * Added new event "JabberStatus" in the Jabber module to monitor buddies
2019 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
2020 in a MixMonitor recording.
2021 * The 'iax2 show peers' output is now similar to the expected output of
2023 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
2025 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
2026 AOC-E messages on a channel.
2027 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
2028 conform more closely to similar events.
2029 * Added a new eventfilter option per user to allow whitelisting and blacklisting
2031 * Added optional parkinglot variable for park command.
2032 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
2033 if CallerIDNum and CallerIDName headers are also present.
2035 Channel Event Logging
2036 ---------------------
2037 * A new interface, CEL, is introduced here. CEL logs single events, much like
2038 the AMI, but it differs from the AMI in that it logs to db backends much
2039 like CDR does; is based on the event subsystem introduced by Russell, and
2040 can share in all its benefits; allows multiple backends to operate like CDR;
2041 is specialized to event data that would be of concern to billing sytems,
2042 like CDR. Backends for logging and accounting calls have been produced,
2043 but a new CDR backend is still in development.
2047 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
2048 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
2049 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
2050 * Multiple files and formats can now be specified in cdr_custom.conf.
2051 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
2052 See configs/cdr_syslog.conf.sample for more information.
2053 * A 'sequence' field has been added to CDRs which can be combined with
2054 linkedid or uniqueid to uniquely identify a CDR.
2055 * Handling of billsec and duration field has changed. If your table definition
2056 specifies those fields as float,double or similar they will now be logged with
2057 microsecond accuracy instead of a whole integer.
2059 Calendaring for Asterisk
2060 ------------------------
2061 * A new set of modules were added supporing calendar integration with Asterisk.
2062 Dialplan functions for reading from and writing to calendars are included,
2063 as well as the ability to execute dialplan logic upon calendar event notifications.
2064 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
2065 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
2066 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
2067 2003 support does not support forms-based authentication).
2069 Call Completion Supplementary Services for Asterisk
2070 ---------------------------------------------------
2071 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
2072 DAHDI/ISDN supports call completion for the following switch types:
2073 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
2074 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
2076 Multicast RTP Support
2077 ---------------------
2078 * A new RTP engine and channel driver have been added which supports Multicast RTP.
2079 The channel driver can be used with the Page application to perform multicast RTP
2080 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
2081 Type can be either basic or linksys.
2082 Destination is the IP address and port for the RTP packets.
2083 Control address is specific to the linksys type and is used for sending the control
2084 packets unique to them.
2086 Security Events Framework
2087 -------------------------
2088 * Asterisk has a new C API for reporting security events. The module res_security_log
2089 sends these events to the "security" logger level. Currently, AMI is the only
2090 Asterisk component that reports security events. However, SIP support will be
2091 coming soon. For more information on the security events framework, see the
2092 "Asterisk Security Framework" section of the Asterisk wiki at
2093 https://wiki.asterisk.org/wiki/x/wgBQ
2094 * SIP support was added in Asterisk 10
2095 * This API now supports IPv6 addresses
2099 * A technology independent fax frontend (res_fax) has been added to Asterisk.
2100 * A spandsp based fax backend (res_fax_spandsp) has been added.
2101 * The app_fax module has been deprecated in favor of the res_fax module and
2102 the new res_fax_spandsp backend.
2103 * The SendFAX and ReceiveFAX applications now send their log messages to a
2104 'fax' logger level, instead of to the generic logger levels. To see these
2105 messages, the system's logger.conf file will need to direct the 'fax' logger
2106 level to one or more destinations; the logger.conf.sample file includes an
2107 example of how to do this. Note that if the 'fax' logger level is *not*
2108 directed to at least one destination, log messages generated by these
2109 applications will be lost, and that if the 'fax' logger level is directed to
2110 the console, the 'core set verbose' and 'core set debug' CLI commands will
2111 have no effect on whether the messages appear on the console or not.
2115 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
2116 Now, in order to enable transmitting silence during record the transmit_silence
2117 option should be used. transmit_silence_during_record remains a valid option, but
2118 defaults to the behavior of the transmit_silence option.
2119 * Addition of the Unit Test Framework API for managing registration and execution
2120 of unit tests with the purpose of verifying the operation of C functions.
2121 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
2122 XMPP text messages to the remote JID.
2123 * Modules.conf has a new option - "require" - that marks a module as critical for
2124 the execution of Asterisk.
2125 If one of the required modules fail to load, Asterisk will exit with a return
2127 * An 'X' option has been added to the asterisk application which enables #exec support.
2128 This allows #exec to be used in asterisk.conf.
2129 * jabber.conf supports a new option auth_policy that toggles auto user registration.
2130 * A new lockconfdir option has been added to asterisk.conf to protect the
2131 configuration directory (/etc/asterisk by default) during reloads.
2132 * The parkeddynamic option has been added to features.conf to enable the creation
2133 of dynamic parkinglots.
2134 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
2135 the reportalarms config option.
2136 * chan_dahdi supports dialing configuring and dialing by device file name.
2137 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
2138 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
2139 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
2140 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
2141 Handy for the above name-based syntax as it does not depend on
2142 initialization order.
2143 * The Realtime dialplan switch now caches entries for 1 second. This provides a
2144 significant increase in performance (about 3X) for installations using this switchtype.
2145 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
2146 AIS. For more information, please see the Distributed Device State section of the
2147 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
2148 * The addition of G.719 pass-through support.
2149 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
2150 during device configuration.
2151 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
2152 have less than 3 lines on the LCD.
2153 * Realtime now supports database failover. See the sample extconfig.conf for details.
2154 * The addition of improved translation path building for wideband codecs. Sample
2155 rate changes during translation are now avoided unless absolutely necessary.
2156 * The addition of the res_stun_monitor module for monitoring and reacting to network
2157 changes while behind a NAT.
2158 * DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf.
2159 DTMF Valid/Invalid number of hits/misses can be set in dsp.conf.
2160 These allow support for any Administration. Default is AT&T values.
2164 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
2165 optionally accept a filename, to apply the setting only to the code generated from
2166 that source file when Asterisk was built. However, there are some modules in Asterisk
2167 that are composed of multiple source files, so this did not result in the behavior
2168 that users expected. In this version, 'core set debug' and 'core set verbose'
2169 can optionally accept *module* names instead (with or without the .so extension),
2170 which applies the setting to the entire module specified, regardless of which source
2171 files it was built from.
2172 * New 'manager show settings' command showing the current settings loaded from
2174 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
2175 the channel hangup request to all channels.
2176 * Added a "core reload" CLI command that executes a global reload of Asterisk.
2178 ------------------------------------------------------------------------------
2179 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
2180 ------------------------------------------------------------------------------
2184 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
2185 Snom phones use this for call pickup of extensions that the phone is
2187 * Added support for setting the domain in the URI for caller of an
2188 outbound call by using the SIPFROMDOMAIN channel variable.
2189 * Added a new configuration option "remotesecret" for authentication to
2190 remote services. For backwards compatibility, "secret" still has the
2191 same function as before, but now you can configure both a remote secret and a
2192 local secret for mutual authentication.
2193 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
2194 the sound will be played to the target of an attended transfer
2195 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
2196 finer control over how many peers Asterisk will qualify and the gap between them
2197 when all peers need to be qualified at the same time.
2198 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
2199 (either globally or for a specific peer), chan_sip will treat any SDP data
2200 it receives as new data and update the media stream accordingly. By
2201 default, Asterisk will only modify the media stream if the SDP session
2202 version received is different from the current SDP session version. This
2203 option is required to interoperate with devices that have non-standard SDP
2204 session version implementations (observed with Microsoft OCS). This option
2205 is disabled by default.
2206 * The parsing of register => lines in sip.conf has been modified to allow a port
2207 to be present in the "user" portion. Please see the sip.conf.sample file for more
2209 * Added support for subscribing to MWI on a remote server and making the status available
2210 as a mailbox. Please see the sip.conf.sample file for more information.
2211 * Added a function to remove SIP headers added in the dialplan before the
2212 first INVITE is generated - SIPRemoveHeader()
2213 * Channel variables set with setvar= in a device configuration is now
2214 set both for inbound and outbound calls.
2215 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
2219 * Added immediate option to iax.conf
2220 * Added forceencryption option to iax.conf
2221 * Added Encryption and Trunk status to manager command "iaxpeers"
2225 * The configuration file now holds separate sections for devices and lines.
2226 Please have a look at configs/skinny.conf.sample and change your skinny.conf
2231 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
2232 support for LibOpenR2. http://www.libopenr2.org/
2233 * The UK option waitfordialtone has been added for use with BT analog
2235 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
2236 is used in conjunction with the 'faxdetect' configuration option. When
2237 'faxbuffers' is used and fax tones are detected, the channel will dynamically
2238 switch to the configured faxbuffers policy. For example, to use 6 buffers
2239 and a 'full' buffer policy for a fax transmission, add:
2241 The faxbuffers configuration will be in affect until the call is torn down.
2242 * Added service message support for 4ESS/5ESS switches.
2246 * For DAHDI channels, the CHANNEL() dialplan function now
2247 supports changing the channel's buffer policy (for the current
2248 call only), using this syntax:
2250 exten => s,n,Set(CHANNEL(buffers)=6,full)
2252 This would change the channel to the 'full' buffer policy and
2253 6 (six) buffers. Possible options for this setting are the same
2254 as those in chan_dahdi.conf.
2255 * Added a new dialplan function, CURLOPT, which permits setting various
2256 options that may be useful with the CURL dialplan function, such as
2257 cookies, proxies, connection timeouts, passwords, etc.
2258 * Permit the syntax and synopsis fields of the corresponding dialplan
2259 functions to be individually set from func_odbc.conf.
2260 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
2261 * func_odbc now may specify an insert query to execute, when the write query
2262 affects 0 rows (usually indicating that no such row exists).
2263 * Added a new dialplan function, LISTFILTER, which permits removing elements
2264 from a set list, by name. Uses the same general syntax as the existing CUT
2265 and FIELDQTY dialplan functions, which also manage lists.
2266 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
2267 obtaining realtime data from the dialplan.
2268 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
2269 a subroutine when using the GoSub() and Return() applications.
2270 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
2271 of "core show function AUDIOHOOK_INHERIT" from the CLI
2272 * Added AES_ENCRYPT. For information on its use, please see the output
2273 of "core show function AES_ENCRYPT" from the CLI
2274 * Added AES_DECRYPT. For information on its use, please see the output
2275 of "core show function AES_DECRYPT" from the CLI
2276 * func_odbc now supports database transactions across multiple queries.
2280 * Scheduled meetme conferences may now have their end times extended by
2282 * app_authenticate now gives the ability to select a prompt other than
2284 * app_directory now pays attention to the searchcontexts setting in
2285 voicemail.conf and will look through all contexts, if no context is
2286 specified in the initial argument.
2287 * A new application, Originate, has been introduced, that allows asynchronous
2288 call origination from the dialplan.
2289 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
2290 in addition to the setting in the "general" context.
2291 * Added ConfBridge dialplan application which does conference bridges without
2292 DAHDI. For information on its use, please see the output of
2293 "core show application ConfBridge" from the CLI.
2297 * The Asterisk CLI has a new command, "channel redirect", which is similar in
2298 operation to the AMI Redirect action.
2299 * extensions.conf now allows you to use keyword "same" to define an extension
2300 without actually specifying an extension. It uses exactly the same pattern
2301 as previously used on the last "exten" line. For example:
2302 exten => 123,1,NoOp(something)
2303 same => n,SomethingElse()
2304 * musiconhold.conf classes of type 'files' can now use relative directory paths,
2305 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
2306 * All deprecated CLI commands are removed from the sourcecode. They are now handled
2307 by the new clialiases module. See cli_aliases.conf.sample file.
2308 * Times within timespecs are now accurate down to the minute. This is a change
2309 from historical Asterisk, which only provided timespecs rounded to the nearest
2310 even (read: evenly divisible by 2) minute mark.
2311 * The realtime switch now supports an option flag, 'p', which disables searches for
2313 * In addition to a time range and date range, timespecs now accept a 5th optional
2314 argument, timezone. This allows you to perform time checks on alternate
2315 timezones, especially if those daylight savings time ranges vary from your
2316 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
2318 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
2319 give you the correct output for an asterisk box behind nat. It will give you the
2320 externhost and localnet settings.
2321 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
2322 can connect calls in passthrough mode, as well as record and play back files.
2323 * Successful and unsuccessful call pickup can now be alerted through sounds, by
2324 using pickupsound and pickupfailsound in features.conf.
2325 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
2326 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
2327 instead of the /var/run/asterisk.pid where it used to be. This will make
2328 installs as non-root easier to manage.
2333 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
2334 be written; they will no longer be explicitly written.
2336 Asterisk Manager Interface
2337 --------------------------
2338 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
2339 a non-empty value) in your request. If you do this, any pending AMI events will
2340 *not* be included in the response to your request as they would normally, but
2341 will be left in the event queue for the next request you make to retrieve. For
2342 some applications, this will allow you to guarantee that you will only see
2343 events in responses to 'WaitEvent' actions, and can better know when to expect them.
2344 To know whether the Asterisk server supports this header or not, your client can
2345 inspect the first response back from the server to see if it includes this header:
2347 Pragma: SuppressEvents
2349 If this is included, the server supports event suppression.
2351 * Added 4 new Actions to list skinny device(s) and line(s)
2357 LDAP Schema File Additions
2358 --------------------------
2359 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
2360 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
2362 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
2363 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
2364 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
2365 * Removed redundant IPaddr (there's already IPAddress)
2366 - Gives more configuration Flags for SIP-Users available (tested)
2367 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
2368 without extensibleObject (which really should be the last resort); gives
2369 also additional possibilities for LDAP-filter
2371 ------------------------------------------------------------------------------
2372 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
2373 ------------------------------------------------------------------------------
2375 Device State Handling
2376 ---------------------
2377 * The event infrastructure in Asterisk got another big update to help support
2378 distributed events. It currently supports distributed device state and
2379 distributed Voicemail MWI (Message Waiting Indication). A new module has
2380 been merged, res_ais, which facilitates communicating events between servers.
2381 It uses the SAForum AIS (Service Availability Forum Application Interface
2382 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
2383 a cluster of Asterisk servers, and to share events between them. For more
2384 information on setting this up, refer to the Distributed Device State section
2385 of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
2389 * Added a new dialplan function, AST_CONFIG(), which allows you to access
2390 variables from an Asterisk configuration file.
2391 * The JACK_HOOK function now has a c() option to supply a custom client name.
2392 * Added two new dialplan functions from libspeex for audio gain control and
2393 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
2394 rx directions of a channel from the dialplan.
2395 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
2396 based on other parameters. The default is still to search based on the
2397 forwarding station ID. However, there are new options that allow you to search
2398 based on the message desk terminal ID, or the message desk number.
2399 * TIMEOUT() has been modified to be accurate down to the millisecond.
2400 * ENUM*() functions now include the following new options:
2401 - 'u' returns the full URI and does not strip off the URI-scheme.
2402 - 's' triggers ISN specific rewriting
2403 - 'i' looks for branches into an Infrastructure ENUM tree
2404 - 'd' for a direct DNS lookup without any flipping of digits.
2405 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
2406 * CHANNEL() now has options for the maximum, minimum, and standard or normal
2407 deviation of jitter, rtt, and loss for a call using chan_sip.
2409 DAHDI channel driver (chan_dahdi) Changes
2410 ----------------------------------------
2411 * Channels can now be configured using named sections in chan_dahdi.conf, just
2412 like other channel drivers, including the use of templates.
2413 * The default for pridialplan has changed from 'national' to 'unknown'.
2417 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
2418 to something that matches the pattern a hint will be created using the contents
2419 and variables evaluated.
2420 * Dialplan matching has been extended to allow an extension to return to the
2421 PBX core to wait for more digits. This is done by using the new dialplan
2422 application called "Incomplete". This will permit a whole new level of
2423 extension control, by giving the administrator more control over early
2424 matches employing one of the short-circuit pattern match operators. Note
2425 that custom applications can trigger this same behavior by returning the
2426 special value AST_PBX_INCOMPLETE.
2430 * Directory now permits both first and last names to be matched at the same
2431 time. In addition, the number of digits to enter of the name can be set in
2432 the arguments to Directory; previously, you could enter only 3, regardless
2433 of how many names are in your company. For large companies, this should be
2435 * Voicemail now permits a mailbox setting to wrap around from first to last
2436 messages, if the "messagewrap" option is set to a true value.
2437 * Voicemail now permits an external script to be run, for password validation.
2438 The script should output "VALID" or "INVALID" on stdout, depending upon the
2439 wish to validate or invalidate the password given. Arguments are:
2440 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
2442 * Dial has a new option: F(context^extension^pri), which permits a callee to
2443 continue in the dialplan, at the specified label, if the caller hangs up.
2444 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
2445 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
2446 * The Jack application now has a c() option to supply a custom client name.
2447 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
2448 like the pre-existing whisper mode, except that the spy can also talk to the
2449 participant on the bridged channel as well.
2450 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
2451 to be spoken instead of the channel name or number. For more information on the
2452 use of this option, issue the command "core show application ChanSpy" from the
2454 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
2455 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
2456 words, if using the 'd' option, it is not possible to enter a number to append to
2457 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
2458 change to whisper mode, and pressing 6 will change to barge mode.
2459 * ExternalIVR now takes several options that affect the way it performs, as
2460 well as having several new commands. Please see the External IVR page on the Asterisk
2461 wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
2462 * Added ability to communicate over a TCP socket instead of forking a child process for the
2463 ExternalIVR application.
2464 * ChanIsAvail has a new option, 'a', which will return all available channels instead
2465 of just the first one if you give the function more then one channel to check.
2466 * PrivacyManager now takes an option where you can specify a context where the
2467 given number will be matched. This way you have more control over who is allowed
2468 and it stops the people who blindly enter 10 digits.
2469 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
2470 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
2471 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
2472 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
2473 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
2474 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
2475 * The Dial() application no longer copies the language used by the caller to the callee's
2476 channel. If you desire for the caller's channel's language to be used for file playback
2477 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
2478 * SendImage() no longer hangs up the channel on error; instead, it sets the
2479 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
2480 'UNSUPPORTED'. This change makes SendImage() more consistent with other
2482 * Park has a new option, 's', which silences the announcement of the parking space number.
2483 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
2484 invalid input and will be assumed to mean that no timeout is desired.
2488 * Added DNS manager support to registrations for peers referencing peer entries.
2489 DNS manager runs in the background which allows DNS lookups to be run asynchronously
2490 as well as periodically updating the IP address. These properties allow for
2491 better performance as well as recovery in the event of an IP change.
2492 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
2493 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
2494 These changes also provide performance improvements for call setup and tear down.
2495 * Added ability to specify registration expiry time on a per registration basis in
2497 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
2499 * Added t38pt_usertpsource option. See sip.conf.sample for details.
2500 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
2501 * 'sip show peers' and 'sip show users' display their entries sorted in
2502 alphabetical order, as opposed to the order they were in, in the config
2504 * Videosupport now supports an additional option, "always", which always sets
2505 up video RTP ports, even on clients that don't support it. This helps with
2506 callfiles and certain transfers to ensure that if two video phones are
2507 connected, they will always share video feeds.
2511 * Existing DNS manager lookups extended to check for SRV records.
2512 * IAX2 encryption support has been improved to support periodic key rotation
2513 within a call for enhanced security. The option "keyrotate" has been
2514 provided to disable this functionality to preserve backwards compatibility
2515 with older versions of IAX2 that do not support key rotation.
2519 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
2520 data tree based on the given <path>.
2521 * New CLI command "data show providers" that will display all the registered
2523 * New CLI command, "config reload <file.conf>" which reloads any module that
2524 references that particular configuration file. Also added "config list"
2525 which shows which configuration files are in use.
2526 * New CLI commands, "pri show version" and "ss7 show version" that will
2527 display which version of libpri and libss7 are being used, respectively.
2528 A new API call was added so trunk will now have to be compiled against
2529 a versions of libpri and libss7 that have them or it will not know that
2530 these libraries exist.
2531 * The commands "core show globals", "core set global" and "core set chanvar" has
2532 been deprecated in favor of the more semanticly correct "dialplan show globals",
2533 "dialplan set chanvar" and "dialplan set global".
2534 * New CLI command "dialplan show chanvar" to list all variables associated
2535 with a given channel.
2539 * Addresses managed by DNS manager now can check to see if there is a DNS
2540 SRV record for a given domain and will use that hostname/port if present.
2542 AMI - The manager (TCP/TLS/HTTP)
2543 --------------------------------
2544 * The Status command now takes an optional list of variables to display
2545 along with channel status.
2546 * The QueueEntry event now also includes the channel's uniqueid
2550 * res_odbc no longer has a limit of 1023 total possible unshared connections,
2551 as some people were running into this limit. This limit has been increased
2556 * The TRANSFER queue log entry now includes the the caller's original
2557 position in the transferred-from queue.
2558 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
2559 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
2560 as well as an explanation about timeout options in general
2561 * Added a new option - C - for forcing the "answered elsewhere" flag on
2562 cancellation of calls in to members of the queue. This is to avoid the
2563 call to a member of a queue having the call listed as a "missed call".
2567 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
2568 adaptive capabilities. What this means in practical terms is that if your
2569 realtime table lacks critical fields, Asterisk will now emit warnings to
2570 that effect. Also, some of the realtime drivers have the ability (if
2571 configured) to automatically add those columns to the table with the
2572 correct type and length.
2576 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
2577 the 'setvar' option to cause a given audio file to be played upon completion
2578 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
2579 Skinny channels only.
2580 * You can now compile Asterisk against the Hoard Memory Allocator, see the
2581 Hoard page on the Asterisk wiki for more information:
2582 https://wiki.asterisk.org/wiki/x/pQBB
2583 * Config file variables may now be appended to, by using the '+=' append
2584 operator. This is most helpful when working with long SQL queries in
2585 func_odbc.conf, as the queries no longer need to be specified on a single
2587 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
2588 which will add a second to the billsec when the ending
2589 time is set, if the number in the microseconds field of the end time is
2590 greater than the number of microseconds in the answer time. This allows
2591 users to count the 'initiated' seconds in their billing records.
2593 ------------------------------------------------------------------------------
2594 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
2595 ------------------------------------------------------------------------------
2597 AMI - The manager (TCP/TLS/HTTP)
2598 --------------------------------
2599 * Manager has undergone a lot of changes, all of them documented
2600 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
2601 * Manager version has changed to 1.1
2602 * Added a new action 'CoreShowChannels' to list currently defined channels
2603 and some information about them.
2604 * Added a new action 'SIPshowregistry' to list SIP registrations.
2605 * Added TLS support for the manager interface and HTTP server
2606 * Added the URI redirect option for the built-in HTTP server
2607 * The output of CallerID in Manager events is now more consistent.
2608 CallerIDNum is used for number and CallerIDName for name.
2609 * Enable https support for builtin web server.
2610 See configs/http.conf.sample for details.
2611 * Added a new action, GetConfigJSON, which can return the contents of an
2612 Asterisk configuration file in JSON format. This is intended to help
2613 improve the performance of AJAX applications using the manager interface
2615 * SIP and IAX manager events now use "ChannelType" in all cases where we
2616 indicate channel driver. Previously, we used a mixture of "Channel"
2617 and "ChannelDriver" headers.
2618 * Added a "Bridge" action which allows you to bridge any two channels that
2619 are currently active on the system.
2620 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
2621 the voicemail users setup.
2622 * Added 'DBDel' and 'DBDelTree' manager commands.
2623 * cdr_manager now reports events via the "cdr" level, separating it from
2624 the very verbose "call" level.
2625 * Manager users are now stored in memory. If you change the manager account
2626 list (delete or add accounts) you need to reload manager.
2627 * Added Masquerade manager event for when a masquerade happens between
2629 * Added "manager reload" command for the CLI
2630 * Lots of commands that only provided information are now allowed under the
2631 Reporting privilege, instead of only under Call or System.
2632 * The IAX* commands now require either System or Reporting privilege, to
2633 mirror the privileges of the SIP* commands.
2634 * Added ability to retrieve list of categories in a config file.
2635 * Added ability to retrieve the content of a particular category.
2636 * Added ability to empty a context.
2637 * Created new action to create a new file.
2638 * Updated delete action to allow deletion by line number with respect to category.
2639 * Added new action insert to add new variable to category at specified line.
2640 * Updated action newcat to allow new category to be inserted in file above another
2642 * Added new event "JitterBufStats" in the IAX2 channel
2643 * Originate now requires the Originate privilege and, if you want to call out
2644 to a subshell, it requires the System privilege, as well. This was done to
2645 enhance manager security.
2646 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
2647 * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
2648 or manager show command Atxfer from the CLI
2649 * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
2650 details or manager show command IAXregistry from the CLI
2654 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
2655 state in the dialplan, as well as creating custom device states that are
2656 controllable from the dialplan.
2657 * Extend CALLERID() function with "pres" and "ton" parameters to
2658 fetch string representation of calling number presentation indicator
2659 and numeric representation of type of calling number value.
2660 * MailboxExists converted to dialplan function
2661 * A new option to Dial() for telling IP phones not to count the call
2662 as "missed" when dial times out and cancels.
2663 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
2664 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
2665 held for any given channel. Also, locks are automatically freed when a
2667 * Added HINT() dialplan function that allows retrieving hint information.
2668 Hints are mappings between extensions and devices for the sake of
2669 determining the state of an extension. This function can retrieve the list
2670 of devices or the name associated with a hint.
2671 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
2673 * Added SYSINFO() dialplan function which allows retrieval of system information
2674 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
2675 the existence of a dialplan target.
2676 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
2677 upper and lower case, respectively.
2678 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
2679 ID for the call (not the Asterisk call ID or unique ID), provided that the
2680 channel driver supports this. For SIP, you get the SIP call-ID for the
2681 bridged channel which you can store in the CDR with a custom field.
2685 * Added CLI permissions, config file: cli_permissions.conf
2686 default is to allow all commands for every local user/group.
2687 Also this new feature added three new CLI commands:
2688 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
2689 - cli reload permissions
2690 - cli show permissions
2691 * New CLI command "core show hint" (usage: core show hint <exten>)
2692 * New CLI command "core show settings"
2693 * Added 'core show channels count' CLI command.
2694 * Added the ability to set the core debug and verbose values on a per-file basis.
2695 * Added 'queue pause member' and 'queue unpause member' CLI commands
2696 * Ability to set process limits ("ulimit") without restarting Asterisk
2697 * Enhanced "agi debug" to print the channel name as a prefix to the debug
2698 output to make debugging on busy systems much easier.
2699 * New CLI commands "dialplan set extenpatternmatching true/false"
2700 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
2701 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
2702 listed in the startup_commands section of cli.conf will get executed.
2703 * Added a CLI command, "devstate change", which allows you to set custom device
2704 states from the func_devstate module that provides the DEVICE_STATE() function
2705 and handling of the "Custom:" devices.
2706 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
2707 sorted into the different possible callbacks, with the number of entries
2708 currently scheduled for each. Gives you a feel for how busy the sip channel
2710 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
2711 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
2712 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
2716 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
2717 option is enabled, Asterisk will watch for a CNG tone in the incoming audio
2718 for a received call. If it is detected, the channel will jump to the
2719 'fax' extension in the dialplan.
2720 * The default SIP useragent= identifier now includes the Asterisk version
2721 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
2722 If set, and the incoming request carries authentication info,
2723 the username to match in the users list is taken from the Digest header
2724 rather than from the From: field. This feature is considered experimental.
2725 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
2726 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
2727 * The "localmask" setting was removed in version 1.2 and the reminder about it
2728 being removed is now also removed.
2729 * A new option "busylevel" for setting a level of calls where asterisk reports
2730 a device as busy, to separate it from call-limit. This value is also added
2731 to the SIP_PEER dialplan function.
2732 * A new realtime family called "sipregs" is now supported to store SIP registration
2733 data. If this family is defined, "sippeers" will be used for configuration and
2734 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
2735 registration data, as before.
2736 * The SIPPEER function have new options for port address, call and pickup groups
2737 * Added support for T.140 realtime text in SIP/RTP
2738 * The "checkmwi" option has been removed from sip.conf, as it is no longer
2739 required due to the restructuring of how MWI is handled. See the descriptions
2740 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
2741 for more information.
2742 * Added rtpdest option to CHANNEL() dialplan function.
2743 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
2744 * SIP now adds a header to the CANCEL if the call was answered by another phone
2745 in the same dial command, or if the new c option in dial() is used.
2746 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
2747 states it is not needed. For phones, however, that do require it the "registertrying" option
2748 has been added so it can be enabled.
2749 * A new option called "callcounter" (global/peer/user level) enables call counters needed
2750 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
2751 used to enable this functionality).
2752 * New settings for timer T1 and timer B on a global level or per device. This makes it
2753 possible to force timeout faster on non-responsive SIP servers. These settings are
2754 considered advanced, so don't use them unless you have a problem.
2755 * Added a dial string option to be able to set the To: header in an INVITE to any
2757 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
2758 the qualify frequency.
2759 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
2760 were not properly torn down due to network or endpoint failures during an established
2762 * Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB
2763 and configs/sip.conf.sample for more information on how it is used.
2764 * Added a new configuration option "authfailureevents" that enables manager events when
2765 a peer can't authenticate properly.
2766 * Added DNS manager support to registrations for peers not referencing a peer entry.
2770 * Added the trunkmaxsize configuration option to chan_iax2.
2771 * Added the srvlookup option to iax.conf
2772 * Added support for OSP. The token is set and retrieved through the CHANNEL()
2775 XMPP Google Talk/Jingle changes
2776 -------------------------------
2777 * Added the bindaddr option to gtalk.conf.
2781 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
2782 * Proper codec support in chan_skinny.
2783 * Added settings for IP and Ethernet QoS requests
2787 * Added separate settings for media QoS in mgcp.conf
2789 Console Channel Driver changes
2790 ------------------------------
2791 * Added experimental support for video send & receive to chan_oss.
2792 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
2795 Phone channel changes (chan_phone)
2796 ----------------------------------
2797 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
2799 H.323 channel Changes
2800 ---------------------
2801 * H323 remote hold notification support added (by NOTIFY message
2802 and/or H.450 supplementary service)
2804 Local channel changes
2805 ---------------------
2806 * The device state functionality in the Local channel driver has been updated
2807 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
2808 to just UNKNOWN if the extension exists.
2809 * Added jitterbuffer support for chan_local. This allows you to use the
2810 generic jitterbuffer on incoming calls going to Asterisk applications.
2811 For example, this would allow you to use a jitterbuffer for an incoming
2812 SIP call to Voicemail by putting a Local channel in the middle. This
2813 feature is enabled by using the 'j' option in the Dial string to the Local
2814 channel in conjunction with the existing 'n' option for local channels.
2815 * A 'b' option has been added which causes chan_local to return the actual channel
2816 that is behind it when queried. This is useful for transfer scenarios as the
2817 actual channel will be transferred, not the Local channel.
2819 Agent channel changes
2820 ----------------------
2821 * The ackcall and endcall options are now supplemented with options acceptdtmf
2822 and enddtmf. These allow for the DTMF keypress to be configurable. The options
2823 default to their old hard-coded values ('#' and '*' respectively) so this should
2824 not break any existing agent installations.
2826 DAHDI channel driver (chan_dahdi) Changes
2827 ----------------------------------------
2828 * SS7 support (via libss7 library)
2829 * In India, some carriers transmit CID via dtmf. Some code has been added
2830 that will handle some situations. The cidstart=polarity_IN choice has been added for
2831 those carriers that transmit CID via dtmf after a polarity change.
2832 * CID matching information is now shown when doing 'dialplan show'.
2833 * Added dahdi show version CLI command.
2834 * Added setvar support to chan_dahdi.conf channel entries.
2835 * Added two new options: mwimonitor and mwimonitornotify. These options allow
2836 you to enable MWI monitoring on FXO lines. When the MWI state changes,
2837 the script specified in the mwimonitornotify option is executed. An internal
2838 event indicating the new state of the mailbox is also generated, so that
2839 the normal MWI facilities in Asterisk work as usual.
2840 * Added signalling type 'auto', which attempts to use the same signalling type
2841 for a channel as configured in DAHDI. This is primarily designed for analog
2842 ports, but will also work for digital ports that are configured for FXS or FXO
2843 signalling types. This mode is also the default now, so if your chan_dahdi.conf
2844 does not specify signalling for a channel (which is unlikely as the sample
2845 configuration file has always recommended specifying it for every channel) then
2846 the 'auto' mode will be used for that channel if possible.
2847 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
2848 state for a channel; also ensured that the DNDState Manager event is
2849 emitted no matter how the DND state is set or cleared.
2853 * Added a new channel driver, chan_unistim. See the Asterisk wiki at
2854 https://wiki.asterisk.org/wiki/x/vgsiAQ and configs/unistim.conf.sample
2855 for details. This new channel driver allows you to use Nortel i2002,
2856 i2004, and i2050 phones with Asterisk.
2857 * Added a new channel driver, chan_console, which uses portaudio as a cross
2858 platform audio interface. It was written as a channel driver that would
2859 work with Mac CoreAudio, but portaudio supports a number of other audio
2860 interfaces, as well. Note that this channel driver requires v19 or higher
2861 of portaudio; older versions have a different API.
2865 * Added the ability to specify arguments to the Dial application when using
2866 the DUNDi switch in the dialplan.
2867 * Added the ability to set weights for responses dynamically. This can be
2868 done using a global variable or a dialplan function. Using the SHELL()
2869 function would allow you to have an external script set the weight for
2871 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
2872 functions will allow you to initiate a DUNDi query from the dialplan,
2873 find out how many results there are, and access each one.
2874 * Added the ability to specifiy a port for a dundi peer.
2878 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
2879 functions will allow you to initiate an ENUM lookup from the dialplan,
2880 and Asterisk will cache the results. ENUMRESULT can be used to access
2881 the results without doing multiple DNS queries.
2885 * Added the ability to customize which sound files are used for some of the
2886 prompts within the Voicemail application by changing them in voicemail.conf
2887 * Added the ability for the "voicemail show users" CLI command to show users
2888 configured by the dynamic realtime configuration method.
2889 * MWI (Message Waiting Indication) handling has been significantly
2890 restructured internally to Asterisk. It is now totally event based
2891 instead of polling based. The voicemail application will notify other
2892 modules that have subscribed to MWI events when something in the mailbox
2894 This also means that if any other entity outside of Asterisk is changing
2895 the contents of mailboxes, then the voicemail application still needs to
2896 poll for changes. Examples of situations that would require this option
2897 are web interfaces to voicemail or an email client in the case of using
2898 IMAP storage. So, two new options have been added to voicemail.conf
2899 to account for this: "pollmailboxes" and "pollfreq". See the sample
2900 configuration file for details.
2901 * Added "tw" language support
2902 * Added support for storage of greetings using an IMAP server
2903 * Added ability to customize forward, reverse, stop, and pause keys for message playback
2904 * SMDI is now enabled in voicemail using the smdienable option.
2905 * A "lockmode" option has been added to asterisk.conf to configure the file
2906 locking method used for voicemail, and potentially other things in the
2907 future. The default is the old behavior, lockfile. However, there is a
2908 new method, "flock", that uses a different method for situations where the
2909 lockfile will not work, such as on SMB/CIFS mounts.
2910 * Added the ability to backup deleted messages, to ease recovery in the case
2911 that a user accidentally deletes a message, and discovers that they need it.
2912 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
2913 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
2914 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
2915 voicemail boxes. The SMDI interface can also poll for MWI changes when some
2916 outside entity is modifying the state of the mailbox (such as IMAP storage or
2917 a web interface of some kind).
2918 * Added the support for marking messages as "urgent." There are two methods to accomplish
2919 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
2920 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
2921 the message as urgent after he has recorded a voicemail by following the voice instructions.
2922 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
2927 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
2928 used across multiple queues.
2929 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
2930 setqueueentryvar options for each queue, see queues.conf.sample for details.
2931 * Added keepstats option to queues.conf which will keep queue
2932 statistics during a reload.
2933 * setinterfacevar option in queues.conf also now sets a variable
2934 called MEMBERNAME which contains the member's name.
2935 * Added 'Strategy' field to manager event QueueParams which represents
2936 the queue strategy in use.
2937 * Added option to run macro when a queue member is connected to a caller,
2938 see queues.conf.sample for details.
2939 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
2940 does not count paused queue members as unavailable.
2941 * Added min-announce-frequency option to queues.conf which allows you to control the
2942 minimum amount of time between queue announcements for use when the caller's queue
2943 position changes frequently.
2944 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
2946 * Added ability for non-realtime queues to have realtime members
2947 * Added the "linear" strategy to queues.
2948 * Added the "wrandom" strategy to queues.
2949 * Added new channel variable QUEUE_MIN_PENALTY
2950 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
2951 rules in queuerules.conf. See configs/queuerules.conf.sample for details
2952 * Added a new parameter for member definition, called state_interface. This may be
2953 used so that a member may be called via one interface but have a different interface's
2954 device state reported.
2955 * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
2956 "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
2957 "manager show command QueueReset."
2958 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
2959 specified by the periodic-announce option, then one will be chosen randomly when it is time
2960 to play a periodic announcment
2961 * New configuration options: announce-position now takes two more values in addition to "yes" and
2962 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
2963 announce-position-limit. By setting announce-position to "limit" callers will only have their
2964 position announced if their position is less than what is specified by announce-position-limit.
2965 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
2966 will be told that their are more than announce-position-limit callers waiting.
2967 * Two new queue log events have been added. An ADDMEMBER event will be logged
2968 when a realtime queue member is added and a REMOVEMEMBER event will be logged
2969 when a realtime queue member is removed. Since there is no calling channel associated
2970 with these events, the string "REALTIME" is placed where the channel's unique id
2971 is typically placed.
2972 * The configuration method for the "joinempty" and "leavewhenempty" options has
2973 changed to a comma-separated list of methods of determining member availability
2974 instead of vague terms such as "yes," "loose," "no," and "strict." These old four
2975 values are still accepted for backwards-compatibility, though.
2976 * The average talktime is now calculated on queues. This information is reported via the
2977 CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
2978 and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
2983 * The 'o' option to provide an optimization has been removed and its functionality
2984 has been enabled by default.
2985 * When a conference is created, the UNIQUEID of the channel that caused it to be
2986 created is stored. Then, every channel that joins the conference will have the
2987 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
2988 callers that come and go from long standing conferences.
2989 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
2990 except it does operations on a channel by name, instead of number in a conference.
2991 This is a very useful feature in combination with the 'X' option to ChanSpy.
2992 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
2994 * Added new RealTime functionality to provide support for scheduled conferencing.
2995 This includes optional messages to the caller if they attempt to join before
2996 the schedule start time, or to allow the caller to join the conference early.
2997 Also included is optional support for limiting the number of callers per
2998 RealTime conference.
2999 * Added the S() and L() options to the MeetMe application. These are pretty
3000 much identical to the S() and L() options to Dial(). They let you set
3001 timeouts for the conference, as well as have warning sounds played to
3002 let the caller know how much time is left, and when it is running out.
3003 * Added the ability to do "meetme concise" with the "meetme" CLI command.
3004 This extends the concise capabilities of this CLI command to include
3005 listing all conferences, instead of an addition to the other sub commands
3006 for the "meetme" command.
3007 * Added the ability to specify the music on hold class used to play into the
3008 conference when there is only one member and the M option is used.
3009 * Added MEETME_INFO dialplan function which provides a way to query
3010 various properties of a Meetme conference.
3011 * Added new admin features: *81: Roll call, *82: eject all, *83: mute all,
3012 and *84: record in-conf
3014 Other Dialplan Application Changes
3015 ----------------------------------
3016 * Argument support for Gosub application
3017 * From the to-do lists: straighten out the app timeout args:
3018 Wait() app now really does 0.3 seconds- was truncating arg to an int.
3019 WaitExten() same as Wait().
3020 Congestion() - Now takes floating pt. argument.
3021 Busy() - now takes floating pt. argument.
3022 Read() - timeout now can be floating pt.
3023 WaitForRing() now takes floating pt timeout arg.
3024 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
3025 * Added 's' option to Page application.
3026 * Added an optional timeout argument to the Page application.
3027 * Added 'E', 'V', and 'P' commands to ExternalIVR.
3028 * Added 'o' and 'X' options to Chanspy.
3029 * Added a new dialplan application, Bridge, which allows you to bridge the
3030 calling channel to any other active channel on the system.
3031 * Added the ability to specify a music on hold class to play instead of ringing
3032 for the SLATrunk application.
3033 * The Read application no longer exits the dialplan on error. Instead, it sets
3034 READSTATUS to ERROR, which you can catch and handle separately.
3035 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
3036 of asking for verification of each name, one at a time.
3037 * Privacy() no longer uses privacy.conf, as all options are specifyable as
3038 direct options to the app.
3039 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
3041 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
3042 * The ChannelRedirect application no longer exits the dialplan if the given channel
3043 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
3044 or NOCHANNEL if the given channel was not found.
3045 * The silencethreshold setting that was previously configurable in multiple
3046 applications is now settable globally via dsp.conf.
3048 Music On Hold Changes
3049 ---------------------
3050 * A new option, "digit", has been added for music on hold classes in
3051 musiconhold.conf. If this is set for a music on hold class, a caller
3052 listening to music on hold can press this digit to switch to listening
3053 to this music on hold class.
3054 * Support for realtime music on hold has been added.
3055 * In conjunction with the realtime music on hold, a general section has
3056 been added to musiconhold.conf, its sole variable is cachertclasses. If this
3057 is set, then music on hold classes found in realtime will be cached in memory.
3061 * AEL upgraded to use the Gosub with Arguments instead
3062 of Macro application, to hopefully reduce the problems
3063 seen with the artificially low stack ceiling that
3064 Macro bumps into. Macros can only call other Macros
3065 to a depth of 7. Tests run using gosub, show depths
3066 limited only by virtual memory. A small test demonstrated
3067 recursive call depths of 100,000 without problems.
3068 -- in addition to this, all apps that allowed a macro
3069 to be called, as in Dial, queues, etc, are now allowing
3070 a gosub call in similar fashion.
3071 * AEL now generates LOCAL(argname) declarations when it
3072 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
3073 etc. That makes the arguments local in scope. The user
3074 can define their own local variables in macros, now,
3075 by saying "local myvar=someval;" or using Set() in this
3076 fashion: Set(LOCAL(myvar)=someval); ("local" is now
3078 * utils/conf2ael introduced. Will convert an extensions.conf
3079 file into extensions.ael. Very crude and unfinished, but
3080 will be improved as time goes by. Should be useful for a
3081 first pass at conversion.
3082 * aelparse will now read extensions.conf to see if a referenced
3083 macro or context is there before issueing a warning.
3084 * AEL parser sets a local channel variable ~~EXTEN~~, to
3085 preserve the value of ${EXTEN} thru switch statements.
3086 * New operator in $[...] expressions: the ~~ operator serves
3087 as a concatenation operator. AT THE MOMENT, it is really only
3088 necessary and useful in AEL, especially in if() expressions.
3089 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
3090 any enclosing double-quotes, and evaluate to the value of a
3091 concatenated with the value of b. For example if a is set to
3092 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
3093 evaluate to xyzabc .
3096 Call Features (res_features) Changes
3097 ------------------------------------
3098 * Added the parkedcalltransfers option to features.conf
3099 * Added parkedcallparking option to control one touch parking w/ parking
3101 * Added parkedcallhangup option to control disconnect feature w/ parking
3103 * Added parkedcallrecording option to control one-touch record w/ parking
3105 * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
3106 parkedcalltransfers option support for multiple parking lots.
3107 * Added BRIDGE_FEATURES variable to set available features for a channel
3108 * The built-in method for doing attended transfers has been updated to
3109 include some new options that allow you to have the transferee sent
3110 back to the person that did the transfer if the transfer is not successful.
3111 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
3112 in features.conf.sample.
3113 * Added support for configuring named groups of custom call features in
3114 features.conf. This means that features can be written a single time, and
3115 then mapped into groups of features for different key mappings or easier
3117 * Updated the ParkedCall application to allow you to not specify a parking
3118 extension. If you don't specify a parking space to pick up, it will grab
3119 the first one available.
3120 * Added cli command 'features reload' to reload call features from features.conf
3121 * Moved into core asterisk binary.
3122 * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
3123 * Added the ability for custom parking lots to be configured with their own
3124 parking extension with the parkext option.
3126 Language Support Changes
3127 ------------------------
3128 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
3129 * Added support for the Hungarian language for saying numbers, dates, and times.
3133 * Added SPEECH commands for speech recognition. A complete listing can be found
3135 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
3136 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
3137 does not behave as expected; the native command needs to be used, instead.
3138 * Added the ability to perform SRV lookups on fast AGI calls. To use this
3139 feature, simply use hagi: instead of agi: as the protocol portion
3140 of the URI parameter to the AGI function call in your dial plan. Also note
3141 that specifying a port number in the AGI URI will disable SRV lookups,
3142 even if you use the hagi: protocol.
3143 * No longer support MSG_OOB flag on HANGUP.
3147 * Added rotatestrategy option to logger.conf, along with two new options:
3148 "timestamp" which will use the time to name the logger files instead of
3149 sequence number; and "rotate", which rotates the names of the log files,
3150 similar to the way syslog rotates files.
3151 * Added exec_after_rotate option to logger.conf, which allows a system
3152 command to be run after rotation. This is primarily useful with
3153 rotatestrategy=rotate, to allow a limit on the number of log files kept
3154 and to ensure that the oldest log file gets deleted.
3155 * Added realtime support for the queue log
3159 * The cdr_manager module has a [mappings] feature, like cdr_custom,
3160 to add fields to the manager event from the CDR variables.
3161 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
3162 backend database CDR table. Specifically, additional, non-standard
3163 columns are supported, merely by setting the corresponding CDR variable in
3164 your dialplan. In addition, you may alias any column to another name (for
3165 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
3166 simply "alias src => ANI" in the configuration file). Records may be
3167 posted to more than one backend, simply by specifying multiple categories
3168 in the configuration file. And finally, you may filter which CDRs get
3169 posted to each backend, by specifying a filter (which the record must
3170 match) for the particular category. Filters are additive (meaning all
3171 rules must match to post that CDR).
3172 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
3173 module. Specifically, you may add additional columns into the table and
3174 they will be set, if you set the corresponding CDR variable name. Also,
3175 if you omit columns in your database table, they will be silently skipped
3176 (but a record will still be inserted, based on what columns remain). Note
3177 that the other two features from cdr_adaptive_odbc (alias and filter) are
3178 not currently supported.
3179 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
3180 has been disabled using the NoCDR application.
3182 Miscellaneous New Modules
3183 -------------------------
3184 * Added a new CDR module, cdr_sqlite3_custom.
3185 * Added a new realtime configuration module, res_config_sqlite
3186 * Added a new codec translation module, codec_resample, which re-samples
3187 signed linear audio between 8 kHz and 16 kHz to help support wideband
3189 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
3190 based on configuration templates that use Asterisk dialplan function and
3191 variable substitution. It should be possible to create phone profiles and
3192 templates that work for the majority of phones provisioned over http. It
3193 is currently only intended to provision a single user account per phone.
3194 An example profile and set of templates for Polycom phones is provided.
3195 NOTE: Polycom firmware is not included, but should be placed in
3196 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
3197 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
3198 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
3199 provided; there is a JACK() application, and a JACK_HOOK() function. Both
3200 interfaces create an input and output JACK port. The application makes
3201 these ports the endpoint of the call. The audio coming from the channel
3202 goes out the output port and whatever comes back in on the input port is
3203 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
3204 audiohook on the channel. This lets you run the audio coming from a
3205 channel through JACK, and whatever comes back in is what gets forwarded
3206 on as the channel's audio. This is very useful for building custom
3207 vocoders or doing recording or analysis of the channel's audio in another
3209 * Added a new module, res_config_curl, which permits using a HTTP POST url
3210 to retrieve, create, update, and delete realtime information from a remote
3211 web server. Note that this module requires func_curl.so to be loaded for
3212 backend functionality.
3213 * Added a new module, res_config_ldap, which permits the use of an LDAP
3214 server for realtime data access.
3215 * Added support for writing and running your dialplan in lua using the pbx_lua
3216 module. See configs/extensions.lua.sample for examples of how to do this.
3220 * Ability to use libcap to set high ToS bits when non-root
3221 on Linux. If configure is unable to find libcap then you
3222 can use --with-cap to specify the path.
3223 * Added maxfiles option to options section of asterisk.conf which allows you to specify
3224 what Asterisk should set as the maximum number of open files when it loads.
3225 * Added the jittertargetextra configuration option.
3226 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
3227 configuration files for the IP channel drivers. The new option is "cos".
3228 This information is also documented on the Asterisk wiki at
3229 https://wiki.asterisk.org/wiki/x/EYBG
3230 * When originating a call using AMI or pbx_spool that fails the reason for failure
3231 will now be available in the failed extension using the REASON dialplan variable.
3232 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
3233 It allows you to configure a prefix for auto-monitor recordings.
3234 * A new extension pattern matching algorithm, based on a trie, is introduced
3235 here, that could noticeably speed up mid-sized to large dialplans.
3236 It is NOT used by default, as duplicating the behaviour of the old pattern
3237 matcher is still under development. A config file option, in extensions.conf,
3238 in the [general] section, called "extenpatternmatchingnew", is by default
3239 set to false; setting that to true will force the use of the new algorithm.
3240 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
3241 be used to switch the algorithms at run time.
3242 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
3243 specifying which socket to use to connect to the running Asterisk daemon
3245 * Performance enhancements to the sched facility, which is used in
3246 the channel drivers, etc. Added hashtabs and doubly-linked lists
3247 to speed up deletion; start at the beginning or end of list to
3249 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
3250 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
3251 Added regression tests to the tests/ dir, also.
3252 * Added a refcount trace feature to astobj2 for those trying to balance
3253 object creation, deletion; work, play; space and time. See the
3254 notes in astobj2.h. Also, see utils/refcounter as well, as a
3255 quick way to find unbalanced refcounts in what could be a sea
3256 of objects that were balanced.
3257 * Added logging to 'make update' command. See update.log
3258 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
3259 do not come from the remote party.
3260 * Added the 'n' option to the SpeechBackground application to tell it to not
3261 answer the channel if it has not already been answered.
3262 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
3263 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
3265 * iLBC source code no longer included (see UPGRADE.txt for details)
3266 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
3267 deadlock is detected, a backtrace of the stack which led to the lock calls
3268 will be output to the CLI.
3269 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
3270 the "core show locks" CLI command will give lock information output as well
3271 as a backtrace of the stack which led to the lock calls.
3272 * users.conf now sports an optional alternateexts property, which permits
3273 allocation of additional extensions which will reach the specified user.
3274 * A new option for the configure script, --enable-internal-poll, has been added
3275 for use with systems which may have a buggy implementation of the poll system
3276 call. If you notice odd behavior such as the CLI being unresponsive on remote
3277 consoles, you may want to try using this option. This option is enabled by default
3278 on Darwin systems since it is known that the Darwin poll() implementation has
3282 --------------------
3283 * In addition to timing from DAHDI, there is a new timing module called
3284 res_timing_timerfd. In order to use this, you must be running Linux with
3285 a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
3286 script will be able to tell if you have the requirements. From menuselect, select
3287 res_timing_timerfd from the Resource Modules menu.
projects / asterisk/asterisk.git / blob